From dujinfang at gmail.com Tue Jan 1 04:05:40 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 1 Jan 2013 09:05:40 +0800 Subject: [Freeswitch-users] mod_event_erlang and invalid process type In-Reply-To: References: Message-ID: First, try this init(Node) -> {foo, Node} ! register_event_handler, {event, Node} ! {event, 'ALL'}. register_event_handler will tell FS to send all events back to the *current* process, so you can get all events by using flush(). 2) If you want to coming events send to a dedicated PID, spawn a process and then send the following message in that process {foo, Node} ! register_event_handler, IN your case, you might want this: loop() -> {foo, Node} ! register_event_handler, {event, Node} ! {event, 'ALL'}, receive stop -> ok; _Anything -> io:format("received ~p~n", [_Anything]), loop() end. 3) Anyway, try read the freeswitch.erl in the source tree to understand more. -- Seven Du Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, December 31, 2012 at 9:12 PM, Gerald Weber wrote: > Some more infos: > > Changed the init/1 function to: > > init(Node) -> > {foo, Node} ! register_event_handler, > {foo, Node} ! register_log_handler, > {foo, Node} ! {event, 'ALL'}, > {foo, Node} ! {set_log_level, debug}. > > Now the ?invalid process type? error message disappeared, but i still dont get any events. > > Some erlang console command outputs: > > freeswitch at fstest> erlang listeners > > Listener to test33 at freeswitch.besharp.at (mailto:test33 at freeswitch.besharp.at) with 0 outbound sessions > > freeswitch at fstest> erlang bindings > > No bindings > > freeswitch at fstest> erlang handlers > > Listener test33 at freeswitch.besharp.at (mailto:test33 at freeswitch.besharp.at): > -------------------------------- > CLONE > CHANNEL_CREATE > CHANNEL_DESTROY > CHANNEL_STATE > CHANNEL_CALLSTATE > CHANNEL_ANSWER > CHANNEL_HANGUP > CHANNEL_HANGUP_COMPLETE > CHANNEL_EXECUTE > CHANNEL_EXECUTE_COMPLETE > CHANNEL_HOLD > CHANNEL_UNHOLD > CHANNEL_BRIDGE > CHANNEL_UNBRIDGE > CHANNEL_PROGRESS > CHANNEL_PROGRESS_MEDIA > CHANNEL_OUTGOING > CHANNEL_PARK > CHANNEL_UNPARK > CHANNEL_APPLICATION > CHANNEL_ORIGINATE > CHANNEL_UUID > API > LOG > INBOUND_CHAN > OUTBOUND_CHAN > STARTUP > SHUTDOWN > PUBLISH > UNPUBLISH > TALK > NOTALK > SESSION_CRASH > MODULE_LOAD > MODULE_UNLOAD > DTMF > MESSAGE > PRESENCE_IN > NOTIFY_IN > PRESENCE_OUT > PRESENCE_PROBE > MESSAGE_WAITING > MESSAGE_QUERY > ROSTER > CODEC > BACKGROUND_JOB > DETECTED_SPEECH > DETECTED_TONE > PRIVATE_COMMAND > HEARTBEAT > TRAP > ADD_SCHEDULE > DEL_SCHEDULE > EXE_SCHEDULE > RE_SCHEDULE > RELOADXML > NOTIFY > SEND_MESSAGE > RECV_MESSAGE > REQUEST_PARAMS > CHANNEL_DATA > GENERAL > COMMAND > SESSION_HEARTBEAT > CLIENT_DISCONNECTED > SERVER_DISCONNECTED > SEND_INFO > RECV_INFO > RECV_RTCP_MESSAGE > CALL_SECURE > NAT > RECORD_START > RECORD_STOP > PLAYBACK_START > PLAYBACK_STOP > CALL_UPDATE > FAILURE > SOCKET_DATA > MEDIA_BUG_START > MEDIA_BUG_STOP > CONFERENCE_DATA_QUERY > CONFERENCE_DATA > CALL_SETUP_REQ > CALL_SETUP_RESULT > CUSTOM: > > Thanks > gw > > Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Gerald Weber > Gesendet: Montag, 31. Dezember 2012 13:30 > An: freeswitch-users at lists.freeswitch.org (mailto:freeswitch-users at lists.freeswitch.org) > Betreff: [Freeswitch-users] mod_event_erlang and invalid process type > > Hi all, > > trying to learn erlang i tried to connect to freeswitch using mod_event_erlang and receive some events: > > erlang shell: > > [root at fstest ~]# erl -name test33 at freeswitch.besharp.at (mailto:test33 at freeswitch.besharp.at) -setcookie ClueCon > Erlang R15B03 (erts-5.9.3.1) [source] [64-bit] [smp:2:2] [async-threads:0] [hipe] [kernel-poll:false] > > Eshell V5.9.3.1 (abort with ^G) > (test33 at freeswitch.besharp.at)1 (mailto:test33 at freeswitch.besharp.at)1)> Pid = fsevents:start(). > <0.46.0> > (test33 at freeswitch.besharp.at)2 (mailto:test33 at freeswitch.besharp.at)2)> fsevents:init('freeswitch at freeswitch.besharp.at (mailto:freeswitch at freeswitch.besharp.at)'). > {event,'ALL'} > (test33 at freeswitch.besharp.at)3 (mailto:test33 at freeswitch.besharp.at)3)> > > > Console output: > > 2012-12-31 13:18:36.900506 [DEBUG] handle_msg.c:301 ALL events enabled > 2012-12-31 13:18:36.900506 [DEBUG] handle_msg.c:314 enable event ALL > 2012-12-31 13:18:51.480577 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>) > 2012-12-31 13:18:51.480577 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>) > 2012-12-31 13:18:51.500492 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>) > > Can anyone tell me what i?m doing wrong ? Console log keeps saying invalid process type (until i kill the erlang process) > and i dont receive any events. > > Erlang Version: 5.9.3.1 > Freeswitch: latest HEAD from 2012-12-31 > > The fsevents module: > > -module(fsevents). > -export([start/0, init/1, loop/0]). > > start() -> > Pid = spawn(?MODULE, loop, []), > register(?MODULE,Pid), > Pid. > > init(Node) -> > {foo, Node} ! {event, 'ALL'}. > > loop() -> > receive > stop -> > ok; > _Anything -> > io:format("received ~p~n", [_Anything]), > loop() > end. > > > The communication seems to work basically: > > > (test33 at freeswitch.besharp.at)4 (mailto:test33 at freeswitch.besharp.at)4)> {foo, freeswitch at freeswitch.besharp.at (mailto:freeswitch at freeswitch.besharp.at)} ! {api, status, ""}. > {api,status,[]} > (test33 at freeswitch.besharp.at)5 (mailto:test33 at freeswitch.besharp.at)5)> receive Y -> Y after 1000 -> timeout end. > {ok,"UP 0 years, 0 days, 1 hour, 20 minutes, 52 seconds, 534 milliseconds, 225 microseconds\nFreeSWITCH (Version 1.3.13b git 8859eb0 2012-12-30 19:08:55Z) is ready\n0 session(s) since startup\n0 session(s) - 0 out of max 30 per sec \n1000 session(s) max\nmin idle cpu 0.00/99.00\nCurrent Stack Size/Max 240K/8192K\n"} > (test33 at freeswitch.besharp.at)3 (mailto:test33 at freeswitch.besharp.at)3)> > > Console output: > > freeswitch at fstest> load mod_erlang_event > 2012-12-31 13:17:21.420611 [INFO] mod_enum.c:872 ENUM Reloaded > 2012-12-31 13:17:21.420611 [INFO] switch_time.c:1165 Timezone reloaded 530 definitions > 2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1816 sections 16 > 2012-12-31 13:17:21.420611 [CONSOLE] switch_loadable_module.c:1348 Successfully Loaded [mod_erlang_event] > 2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:254 Adding Application 'erlang' > 2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:254 Adding Application 'erlang_sendmsg' > 2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:298 Adding API Function 'erlang' > > +OK Reloading XML > +OK > > 2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1915 Socket 48 up listening on 0.0.0.0:8031 > 2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1946 Connected to epmd and published erlang cnode at freeswitch at freeswitch.besharp.at (mailto:freeswitch at freeswitch.besharp.at) > freeswitch at fstest> 2012-12-31 13:17:43.380527 [DEBUG] mod_erlang_event.c:1986 Launching listener, connection from node test33 at freeswitch.besharp.at (mailto:test33 at freeswitch.besharp.at), ip 192.168.20.73 > 2012-12-31 13:17:43.380527 [DEBUG] mod_erlang_event.c:1009 Connection Open from 192.168.20.73 > > Thanks / regards and Happy New Year J > Gw > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130101/01beee9d/attachment-0001.html From Sirish.MasurMohan at oa.com.au Wed Jan 2 04:28:52 2013 From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan) Date: Wed, 2 Jan 2013 12:28:52 +1100 Subject: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled? In-Reply-To: References: <965759A53E43FE439E43565A7715E5F058F4156D1E@oa-exchange1.oa.com.au> <50CE5D6E.2020401@quentustech.com> <965759A53E43FE439E43565A7715E5F058F4156D32@oa-exchange1.oa.com.au> <965759A53E43FE439E43565A7715E5F058F4156DD8@oa-exchange1.oa.com.au> <965759A53E43FE439E43565A7715E5F058F4156DF7@oa-exchange1.oa.com.au> <965759A53E43FE439E43565A7715E5F058F4156E6C@oa-exchange1.oa.com.au> <849EC2AD-D5F4-471E-B362-480DF549DB62@gmail.com> Message-ID: <965759A53E43FE439E43565A7715E5F058F4156ED7@oa-exchange1.oa.com.au> Hi Michael, I have modified my implementation to use hash, as recommended. I now use a Lua script which will * hash selects to read the port status & previous timestamp of each port * decide on the port to be used based on the oldest time stamp - hash insert the status of this port to busy On hang-up, I have another Lua script which would hash insert the status of this port back to idle, and hash insert the latest timestamp If I were to have two or more incoming calls at the same time, and the answering Lua script were to get executed at the same time for these calls, I am still worried about the concurrency factor - wouldn't the Lua script hash select and return the same port? If yes, how do I 'protect' this script to handle concurrency? Thanks! With regards, Sirish From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, 22 December 2012 4:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled? Agreed, the hash/db options are safe and effective. You don't need global variables. I highly recommend that you look at the default.xml dialplan file and look at the places where "hash" occurs. In Local_Extension a lot of items are written to the local database with hash "inserts" and elsewhere you'll see hash "selects" and the occasional hash "delete". Use the hash_dump command to see what's in the database. Make a few test calls and do a hash_dump before, during, and after the call. You'll quickly see what you can do with the hash command and storing information. Just store information for each of your ports and use a Lua script to read and/or write as needed. -MC On Thu, Dec 20, 2012 at 6:32 PM, Steven Ayre > wrote: If I remember correctly when when anthm added the option by request, global_setvar is NOT safe for setting concurrently. Very occasionally should be ok for config changes without a restart, but it is not safe to use from call dial plans where calls are running alongside each other. You should look at using mod_hash or mod_db, which are much safer Sent from my iPad On 20 Dec 2012, at 23:23, Sirish Masur Mohan > wrote: Hi Michael, Thanks for the reply, and the suggestions. My current implementation is a similar to what you have suggested, i.e.: 1. I have defined 4 sets of global variables in vars.xml, where each set keeps a track of port's status and last used timestamp 2. In the dialplan, I execute a simple Lua script which 'chooses' the line based on global variables status and timestamp (using "global_getvar") and update the status of the line to be used (using "global_setvar") 3. Bridge to the line returned by the above script 4. Execute another Lua script on hang-up, which updates the global variables (status and timestamp) If I were to implement the hash way, I would have to probably do the following: 1. In the dialplan, I execute a simple Lua script which 'chooses' the line by reading the hash values of the line status and timestamp and update the hash status of the line to be used 2. Bridge to the line returned by the above script 3. Execute another Lua script on hang-up, which updates the hash values (status and timestamp) But I am still not clear on the concurrency question - if FreeSWITCH has received 2 calls at the same time on the E1 line, should I be worried about protecting the script that gets executed to access the hash, decide on the line to be used, and then update the hash? Thanks again! With regards, Sirish From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, 21 December 2012 3:42 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled? Sirish, Since you're using an E1 to connect to the PBX then really all you need to do is keep track of the last time each port was hung up and whether or not a given port is currently in use. I would use api_on_answer to launch a simple script to set a flag to say that a particular port is in use and then use the api_hangup_hook to launch another script when the call ends. The channel variables page on the wiki has some examples of how to use these. I recommend that you write simple Lua scripts that use the "hash" API to store information in the local database. Also, check out the "hash_dump" API as it is a useful way to quickly see what all is stored there. For an example of how to add, remove, and read information from the local database using the "hash" API please see conf/dialplan/default.xml. Search for "hash" and you'll see all sorts of examples of how the example dialplan uses the local database to store useful information that allows us to implement features like call return, call intercept, etc. -MC On Tue, Dec 18, 2012 at 7:39 PM, Sirish Masur Mohan > wrote: Hi Michael, >> How are you physically connecting from FreeSWITCH to the PBX? I connect this via E1 link - I have a Sangoma card installed on the FreeSWITCH machine. With regards, Sirish From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, 19 December 2012 1:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled? To trigger SIP registrations you'd need the PBX to have a SIP client. I'm assuming this is possible, but maybe that's a false assumption. How are you physically connecting from FreeSWITCH to the PBX? -MC On Tue, Dec 18, 2012 at 2:36 PM, Sirish Masur Mohan > wrote: Hey Guys, Would really appreciate if you could help me out here - isn't there a way to handle concurrent calls in the dial plan, especially when Lua scripts, accessing global variables, are executed on receiving calls? Is mod_fifo the closest I could get to handle concurrency (as Michael has explained)? If yes, how do I trigger SIP registrations, especially working with a PBX which I don't have full control of? With regards, Sirish From: Sirish Masur Mohan Sent: Tuesday, 18 December 2012 1:02 PM To: FreeSWITCH Users Help Subject: RE: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled? Hi Michael, Thanks for the reply. >> You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions I am seen SIP clients such as X-Lite sending out the SIP registrations, but could you please clarify as to how this can be achieved in the PBX? The final production environment would be out in the customer's PBX, which I may not have complete control of.. >> What application are you building? I may not be able to provide the details because of the NDA with customer, but what I am trying to achieve is, to replace an existing IVR with FreeSWITCH in an old existing setup of the customer - that's the reason why we continue working with dialup modems! With regards, Sirish From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, 18 December 2012 6:23 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled? You don't have to have actual human agents for mod_fifo. You could define a user for each modem and then manually "log in" those "agents" on the command line using the fifo_member API command. Something like this: fifo_member add fifo_name {fifo_member_wait=nowait}user/1234 Where 1234 is the user id of one of the modems. You would need a SIP registration from the PBX to FreeSWITCH for each of the modem extensions. Having modems go through a VoIP system sounds a bit scary. What application are you building? -MC On Sun, Dec 16, 2012 at 5:41 PM, Sirish Masur Mohan > wrote: Hi William, Thanks for the reply. My setup is as follows: Client(Caller)->dialup modem->PBX->FreeSWITCH->PBX-> 4 dialup modems->Server(Receiver) I basically need FreeSWITCH to bridge the incoming call to the best external destination (out of the 4 available), so that the modem training, connection etc can take place smoothly, before exchange of data. I am not sure if mod_fifo would help me in this scenario, as, I would require an agent to dial in and read the fifo. Could you please clarify? Thanks! With regards, Sirish -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King Sent: Monday, 17 December 2012 10:47 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie question: Executing Lua scripts for incoming calls, how concurrency is to be handled? Sounds like you want to take a look into mod_fifo. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/4c0f7ed2/attachment-0001.html From babak.freeswitch at gmail.com Wed Jan 2 07:54:43 2013 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Wed, 2 Jan 2013 08:24:43 +0330 Subject: [Freeswitch-users] registering on asterisk getting 403 Forbidden (Bad auth) Message-ID: hi this is my gateway: this is created on a profile exactly like external but on port 5090. target is an asterisk box. packets are in: http://pastebin.freeswitch.org/20375 other party says that is getting username = null in their logs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/e8502e32/attachment.html From sanjay.sanjaysoni at gmail.com Wed Jan 2 09:26:28 2013 From: sanjay.sanjaysoni at gmail.com (sanjay soni) Date: Wed, 2 Jan 2013 11:56:28 +0530 Subject: [Freeswitch-users] Accept google talk friend request in dingaling-client mode Message-ID: Hello, I have client set-up of mod-dingaling and am able to get gtalk/GV calls working both sides (inward and outward). I am controlling the FS from my app using Python ESL client and static dialplan. However, new accounts can not talk to each other unless they are already friends. Is there a way to get mod-dingaling to automatically accept incoming friend requests. I dont even see a log to that effect. I know that adding/accepting friend request works in component mode But I specifically want to know about client mode. Whether - A) This feature can be added in the next upcoming versions or based on the contract B) This can not happen because of the way google jingle - FS dingaling works in client mode. Thanks Sanjay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/4b157e25/attachment.html From engineerzuhairraza at gmail.com Wed Jan 2 12:11:46 2013 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Wed, 2 Jan 2013 13:11:46 +0400 Subject: [Freeswitch-users] Accept google talk friend request in dingaling-client mode In-Reply-To: References: Message-ID: I used this bot to accept requests *http://code.google.com/p/pygtalkrobot/* * * Regards, Zohair Raza On Wed, Jan 2, 2013 at 10:26 AM, sanjay soni wrote: > Hello, > I have client set-up of mod-dingaling and am able to get gtalk/GV calls > working both sides (inward and outward). I am controlling the FS from my > app using Python ESL client and static dialplan. > However, new accounts can not talk to each other unless they are already > friends. Is there a way to get mod-dingaling to automatically accept > incoming friend requests. I dont even see a log to that effect. I know that > adding/accepting friend request works in component mode But I specifically > want to know about client mode. Whether - A) This feature can be added in > the next upcoming versions or based on the contract B) This can not happen > because of the way google jingle - FS dingaling works in client mode. > > Thanks > Sanjay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/c55abe53/attachment.html From gerald.weber at besharp.at Wed Jan 2 13:27:10 2013 From: gerald.weber at besharp.at (Gerald Weber) Date: Wed, 2 Jan 2013 10:27:10 +0000 Subject: [Freeswitch-users] mod_event_erlang and invalid process type In-Reply-To: References: Message-ID: Thanks for your answer, it?s working now ? Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Seven Du Gesendet: Dienstag, 01. J?nner 2013 02:06 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] mod_event_erlang and invalid process type First, try this init(Node) -> {foo, Node} ! register_event_handler, {event, Node} ! {event, 'ALL'}. register_event_handler will tell FS to send all events back to the *current* process, so you can get all events by using flush(). 2) If you want to coming events send to a dedicated PID, spawn a process and then send the following message in that process {foo, Node} ! register_event_handler, IN your case, you might want this: loop() -> {foo, Node} ! register_event_handler, {event, Node} ! {event, 'ALL'}, receive stop -> ok; _Anything -> io:format("received ~p~n", [_Anything]), loop() end. 3) Anyway, try read the freeswitch.erl in the source tree to understand more. -- Seven Du Sent with Sparrow On Monday, December 31, 2012 at 9:12 PM, Gerald Weber wrote: Some more infos: Changed the init/1 function to: init(Node) -> {foo, Node} ! register_event_handler, {foo, Node} ! register_log_handler, {foo, Node} ! {event, 'ALL'}, {foo, Node} ! {set_log_level, debug}. Now the ?invalid process type? error message disappeared, but i still dont get any events. Some erlang console command outputs: freeswitch at fstest> erlang listeners Listener to test33 at freeswitch.besharp.at with 0 outbound sessions freeswitch at fstest> erlang bindings No bindings freeswitch at fstest> erlang handlers Listener test33 at freeswitch.besharp.at: -------------------------------- CLONE CHANNEL_CREATE CHANNEL_DESTROY CHANNEL_STATE CHANNEL_CALLSTATE CHANNEL_ANSWER CHANNEL_HANGUP CHANNEL_HANGUP_COMPLETE CHANNEL_EXECUTE CHANNEL_EXECUTE_COMPLETE CHANNEL_HOLD CHANNEL_UNHOLD CHANNEL_BRIDGE CHANNEL_UNBRIDGE CHANNEL_PROGRESS CHANNEL_PROGRESS_MEDIA CHANNEL_OUTGOING CHANNEL_PARK CHANNEL_UNPARK CHANNEL_APPLICATION CHANNEL_ORIGINATE CHANNEL_UUID API LOG INBOUND_CHAN OUTBOUND_CHAN STARTUP SHUTDOWN PUBLISH UNPUBLISH TALK NOTALK SESSION_CRASH MODULE_LOAD MODULE_UNLOAD DTMF MESSAGE PRESENCE_IN NOTIFY_IN PRESENCE_OUT PRESENCE_PROBE MESSAGE_WAITING MESSAGE_QUERY ROSTER CODEC BACKGROUND_JOB DETECTED_SPEECH DETECTED_TONE PRIVATE_COMMAND HEARTBEAT TRAP ADD_SCHEDULE DEL_SCHEDULE EXE_SCHEDULE RE_SCHEDULE RELOADXML NOTIFY SEND_MESSAGE RECV_MESSAGE REQUEST_PARAMS CHANNEL_DATA GENERAL COMMAND SESSION_HEARTBEAT CLIENT_DISCONNECTED SERVER_DISCONNECTED SEND_INFO RECV_INFO RECV_RTCP_MESSAGE CALL_SECURE NAT RECORD_START RECORD_STOP PLAYBACK_START PLAYBACK_STOP CALL_UPDATE FAILURE SOCKET_DATA MEDIA_BUG_START MEDIA_BUG_STOP CONFERENCE_DATA_QUERY CONFERENCE_DATA CALL_SETUP_REQ CALL_SETUP_RESULT CUSTOM: Thanks gw Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Gerald Weber Gesendet: Montag, 31. Dezember 2012 13:30 An: freeswitch-users at lists.freeswitch.org Betreff: [Freeswitch-users] mod_event_erlang and invalid process type Hi all, trying to learn erlang i tried to connect to freeswitch using mod_event_erlang and receive some events: erlang shell: [root at fstest ~]# erl -name test33 at freeswitch.besharp.at -setcookie ClueCon Erlang R15B03 (erts-5.9.3.1) [source] [64-bit] [smp:2:2] [async-threads:0] [hipe] [kernel-poll:false] Eshell V5.9.3.1 (abort with ^G) (test33 at freeswitch.besharp.at)1> Pid = fsevents:start(). <0.46.0> (test33 at freeswitch.besharp.at)2> fsevents:init('freeswitch at freeswitch.besharp.at'). {event,'ALL'} (test33 at freeswitch.besharp.at)3> Console output: 2012-12-31 13:18:36.900506 [DEBUG] handle_msg.c:301 ALL events enabled 2012-12-31 13:18:36.900506 [DEBUG] handle_msg.c:314 enable event ALL 2012-12-31 13:18:51.480577 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>) 2012-12-31 13:18:51.480577 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>) 2012-12-31 13:18:51.500492 [ERR] ei_helpers.c:274 Invalid process type 0! ((null),, <0.0.0>) Can anyone tell me what i?m doing wrong ? Console log keeps saying invalid process type (until i kill the erlang process) and i dont receive any events. Erlang Version: 5.9.3.1 Freeswitch: latest HEAD from 2012-12-31 The fsevents module: -module(fsevents). -export([start/0, init/1, loop/0]). start() -> Pid = spawn(?MODULE, loop, []), register(?MODULE,Pid), Pid. init(Node) -> {foo, Node} ! {event, 'ALL'}. loop() -> receive stop -> ok; _Anything -> io:format("received ~p~n", [_Anything]), loop() end. The communication seems to work basically: (test33 at freeswitch.besharp.at)4> {foo, freeswitch at freeswitch.besharp.at} ! {api, status, ""}. {api,status,[]} (test33 at freeswitch.besharp.at)5> receive Y -> Y after 1000 -> timeout end. {ok,"UP 0 years, 0 days, 1 hour, 20 minutes, 52 seconds, 534 milliseconds, 225 microseconds\nFreeSWITCH (Version 1.3.13b git 8859eb0 2012-12-30 19:08:55Z) is ready\n0 session(s) since startup\n0 session(s) - 0 out of max 30 per sec \n1000 session(s) max\nmin idle cpu 0.00/99.00\nCurrent Stack Size/Max 240K/8192K\n"} (test33 at freeswitch.besharp.at)3> Console output: freeswitch at fstest> load mod_erlang_event 2012-12-31 13:17:21.420611 [INFO] mod_enum.c:872 ENUM Reloaded 2012-12-31 13:17:21.420611 [INFO] switch_time.c:1165 Timezone reloaded 530 definitions 2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1816 sections 16 2012-12-31 13:17:21.420611 [CONSOLE] switch_loadable_module.c:1348 Successfully Loaded [mod_erlang_event] 2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:254 Adding Application 'erlang' 2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:254 Adding Application 'erlang_sendmsg' 2012-12-31 13:17:21.420611 [NOTICE] switch_loadable_module.c:298 Adding API Function 'erlang' +OK Reloading XML +OK 2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1915 Socket 48 up listening on 0.0.0.0:8031 2012-12-31 13:17:21.420611 [DEBUG] mod_erlang_event.c:1946 Connected to epmd and published erlang cnode at freeswitch at freeswitch.besharp.at freeswitch at fstest> 2012-12-31 13:17:43.380527 [DEBUG] mod_erlang_event.c:1986 Launching listener, connection from node test33 at freeswitch.besharp.at, ip 192.168.20.73 2012-12-31 13:17:43.380527 [DEBUG] mod_erlang_event.c:1009 Connection Open from 192.168.20.73 Thanks / regards and Happy New Year ? Gw _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/7e1bcdb8/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Wed Jan 2 15:19:13 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 2 Jan 2013 12:19:13 +0000 Subject: [Freeswitch-users] mod_xml_cdr and mod_json_cdr - docs/code Message-ID: Hello, Looking at the source code for xml_cdr and json_cdr: https://github.com/FreeSWITCH/FreeSWITCH/blob/master/src/mod/event_handlers/mod_json_cdr/mod_json_cdr.c https://github.com/FreeSWITCH/FreeSWITCH/blob/master/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c Based on the commit messages and file contents, it looks like this code is maintained in two different places, and is mostly copy/paste. Looking at the documentation, all the relevant channel information is inside the xml_cdr docs, with very little inside json_cdr, despite them having nearly the same code base. Could a core dev comment on; * Why this code is maintained in two different places - are there any plans to merge it into a single lib/file? * Is there functionally any difference between the two, other than the fact one outputs XML and another outputs JSON (my C is very poor, so I find it difficult to interpret the code logic - sorry) Many thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/ae237bd5/attachment.html From andrew at cassidywebservices.co.uk Wed Jan 2 15:47:56 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 2 Jan 2013 12:47:56 +0000 Subject: [Freeswitch-users] mod_xml_cdr and mod_json_cdr - docs/code In-Reply-To: References: Message-ID: I'm not a core dev, but my guess would be to reduce the risk of breaking current installations while mod_json_cdr was being developed. I do see you point but it does make more sense to move it out. perhaps the common code could be librarified or they could be merged later on, but not breaking current installations is a pretty high priority. This is what happened last time a Linux kernel dev broke something big: http://goo.gl/u8GWl On 2 January 2013 12:19, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello, > > Looking at the source code for xml_cdr and json_cdr: > > > https://github.com/FreeSWITCH/FreeSWITCH/blob/master/src/mod/event_handlers/mod_json_cdr/mod_json_cdr.c > > https://github.com/FreeSWITCH/FreeSWITCH/blob/master/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c > > Based on the commit messages and file contents, it looks like this code is > maintained in two different places, and is mostly copy/paste. > > Looking at the documentation, all the relevant channel information is > inside the xml_cdr docs, with very little inside json_cdr, despite them > having nearly the same code base. > > Could a core dev comment on; > > * Why this code is maintained in two different places - are there any > plans to merge it into a single lib/file? > * Is there functionally any difference between the two, other than the > fact one outputs XML and another outputs JSON (my C is very poor, so I find > it difficult to interpret the code logic - sorry) > > Many thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/2d9236a9/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed Jan 2 16:03:42 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 2 Jan 2013 13:03:42 +0000 Subject: [Freeswitch-users] mod_xml_cdr and mod_json_cdr - docs/code In-Reply-To: References: Message-ID: lmao - that is a pretty funny thread. Just realised JIRA is the best place for this - ticket raised; http://jira.freeswitch.org/browse/FS-4983 Thanks Cal On Wed, Jan 2, 2013 at 12:47 PM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > I'm not a core dev, but my guess would be to reduce the risk of breaking > current installations while mod_json_cdr was being developed. I do see you > point but it does make more sense to move it out. perhaps the common code > could be librarified or they could be merged later on, but not breaking > current installations is a pretty high priority. This is what happened last > time a Linux kernel dev broke something big: http://goo.gl/u8GWl > > On 2 January 2013 12:19, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello, >> >> Looking at the source code for xml_cdr and json_cdr: >> >> >> https://github.com/FreeSWITCH/FreeSWITCH/blob/master/src/mod/event_handlers/mod_json_cdr/mod_json_cdr.c >> >> https://github.com/FreeSWITCH/FreeSWITCH/blob/master/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c >> >> Based on the commit messages and file contents, it looks like this code >> is maintained in two different places, and is mostly copy/paste. >> >> Looking at the documentation, all the relevant channel information is >> inside the xml_cdr docs, with very little inside json_cdr, despite them >> having nearly the same code base. >> >> Could a core dev comment on; >> >> * Why this code is maintained in two different places - are there any >> plans to merge it into a single lib/file? >> * Is there functionally any difference between the two, other than the >> fact one outputs XML and another outputs JSON (my C is very poor, so I find >> it difficult to interpret the code logic - sorry) >> >> Many thanks >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/38970b3a/attachment.html From odermann at googlemail.com Wed Jan 2 17:10:24 2013 From: odermann at googlemail.com (Dennis) Date: Wed, 2 Jan 2013 15:10:24 +0100 Subject: [Freeswitch-users] Need to recompile fs after ODBC version update? Message-ID: hi, i just like to ask, if someone could tell me, if i have to recompile fs after updating the odbc-version on the server? thanks & kind regrads dennis From lists at kavun.ch Wed Jan 2 17:37:55 2013 From: lists at kavun.ch (Emrah) Date: Wed, 2 Jan 2013 09:37:55 -0500 Subject: [Freeswitch-users] Random no audio Message-ID: <4F46684C-9975-468D-B87A-3FAA357AD7AD@kavun.ch> Hi guys, Happy new year to all of you and may all your wishes be fulfilled. I think I may have broken something while trying to fix my codec issues. I randomly get no audio and even though I automatically thought it was a NAT issue, I now tend to think it's only codec related. I've been pulling my hair off this for a while now and your help would be immensely appreciated. Pastebin here: http://pastebin.freeswitch.org/20377 Thank you and all the best, Emrah From steveayre at gmail.com Wed Jan 2 17:48:01 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 2 Jan 2013 14:48:01 +0000 Subject: [Freeswitch-users] Need to recompile fs after ODBC version update? In-Reply-To: References: Message-ID: Unless there has been an ABI change, if it's using the shared library it should only require a restart. -Steve On 2 January 2013 14:10, Dennis wrote: > hi, > > i just like to ask, if someone could tell me, if i have to recompile > fs after updating the odbc-version on the server? > > thanks & kind regrads > dennis > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/89d3cb75/attachment.html From brian at freeswitch.org Wed Jan 2 19:27:10 2013 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Jan 2013 10:27:10 -0600 Subject: [Freeswitch-users] Happy New Year! Message-ID: http://blog.smartbear.com/software-quality/bid/167051/14-Ways-to-Contribute-to-Open-Source-without-Being-a-Programming-Genius-or-a-Rock-Star I need some folks to step up and really help out with Jira... Any takers? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 ISN: 410*543 From kb.santosh.kadam at gmail.com Wed Jan 2 14:05:54 2013 From: kb.santosh.kadam at gmail.com (Santosh Kadam) Date: Wed, 2 Jan 2013 16:35:54 +0530 Subject: [Freeswitch-users] mod_mp4 working and usage Message-ID: Hello there, Can someone explain if the mod_mp4 is working ? If yes, can someone forward the procedure to make the same work ? Currently - I am using FreeSwitch with Xlite 5.0 as my SIP client. Does this client support playback of MP4 video? If not, please suggest an appropriate SIP client as well. thanks in advance for the response. Warm Regards santosh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/5f94e12a/attachment.html From sanjay.sanjaysoni at gmail.com Wed Jan 2 14:28:28 2013 From: sanjay.sanjaysoni at gmail.com (soni4m) Date: Wed, 2 Jan 2013 03:28:28 -0800 (PST) Subject: [Freeswitch-users] Accept google talk friend request in dingaling-client mode In-Reply-To: References: Message-ID: <1357126108411-7586001.post@n2.nabble.com> Thanks, I haven?t used this earlier. I am exploring its capability of notifying me of incoming add contact request and will update here how it worked with my app once I am done. But, just to understand the flow ? You are saying that along with logging in the google ID through mod_dingaling I should also login through this bot. The Bot will handle for any incoming friend request and accept them automatically, meanwhile my jingle-sip business will go on through dingaling. Thanks Sanjay -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Accept-google-talk-friend-request-in-dingaling-client-mode-tp7585998p7586001.html Sent from the freeswitch-users mailing list archive at Nabble.com. From engineerzuhairraza at gmail.com Wed Jan 2 19:39:30 2013 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Wed, 2 Jan 2013 20:39:30 +0400 Subject: [Freeswitch-users] Accept google talk friend request in dingaling-client mode In-Reply-To: <1357126108411-7586001.post@n2.nabble.com> References: <1357126108411-7586001.post@n2.nabble.com> Message-ID: Yes It will work like thia You need to login from both places On Jan 2, 2013 8:37 PM, "soni4m" wrote: > Thanks, I haven?t used this earlier. I am exploring its capability of > notifying me of incoming add contact request and will update here how it > worked with my app once I am done. > > But, just to understand the flow ? You are saying that along with logging > in > the google ID through mod_dingaling I should also login through this bot. > The Bot will handle for any incoming friend request and accept them > automatically, meanwhile my jingle-sip business will go on through > dingaling. > > Thanks > Sanjay > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Accept-google-talk-friend-request-in-dingaling-client-mode-tp7585998p7586001.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/55e46c3d/attachment.html From msc at freeswitch.org Wed Jan 2 20:15:39 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Jan 2013 09:15:39 -0800 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today Message-ID: Hello all! Welcome back from the holidays. We hope everyone enjoyed some time off. With the new year we are starting to get back into the swing of things. Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_02 We are going to discuss some new utility scripts that have been submitted. If you have a utility script or an interesting hangup hook trick then by all means please share it. We will also be populating a new section of the wiki with those hangup hook tricks. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/69c32c41/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed Jan 2 20:30:00 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 2 Jan 2013 17:30:00 +0000 Subject: [Freeswitch-users] 5 for 1 offer - senior/core devs Message-ID: Hello, Sometimes it's difficult to get the attention of the core devs, they have to prioritize like everyone else. One of the approaches used on the Django mailing list was a 5 for 1 offer.. this is where you help out on 5 tickets (usually low priority ones), and a senior/core dev will look at one of your tickets in return (no guarantees of a fix, just purely to get their attention/comments) Would this approach be welcomed by the FS core team? Could be a great way of not only getting more tickets looked at, whilst providing a reward for doing so (as an alternative to bountys) Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/c507d212/attachment-0001.html From mickstevens at yahoo.com Wed Jan 2 20:38:48 2013 From: mickstevens at yahoo.com (Mick Stevens) Date: Wed, 2 Jan 2013 09:38:48 -0800 (PST) Subject: [Freeswitch-users] 5 for 1 offer - senior/core devs In-Reply-To: References: Message-ID: <1357148328.77687.YahooMailNeo@web160802.mail.bf1.yahoo.com> Great idea, count me in please (as one of the junior/amateur helpers)! ? Rgds, Mick Tel/SMS. +44(0)7967 594432 Fax. +44(0)7053 452429 Email/IM. mickstevens at yahoo.com Skype: mick_stevens www.facebook.com/mickstevens www.twitter.com/mickstevens ________________________________ From: Cal Leeming [Simplicity Media Ltd] To: FreeSWITCH Users Help Sent: Wednesday, 2 January 2013, 17:30 Subject: [Freeswitch-users] 5 for 1 offer - senior/core devs Hello, Sometimes it's difficult to get the attention of the core devs, they have to prioritize like everyone else. One of the approaches used on the Django mailing list was a 5 for 1 offer.. this is where you help out on 5 tickets (usually low priority ones), and a senior/core dev will look at one of your tickets in return (no guarantees of a fix, just purely to get their attention/comments) Would this approach be welcomed by the FS core team? Could be a great way of not only getting more tickets looked at, whilst providing a reward for doing so (as an alternative to bountys) Cal _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/ea0765bc/attachment.html From william.king at quentustech.com Wed Jan 2 20:55:17 2013 From: william.king at quentustech.com (William King) Date: Wed, 02 Jan 2013 09:55:17 -0800 Subject: [Freeswitch-users] 5 for 1 offer - senior/core devs In-Reply-To: References: Message-ID: <50E47485.2010107@quentustech.com> It would require someone who was willing to 'manage' the karma system. Any links to how the django mailing list handles it? William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 01/02/2013 09:30 AM, Cal Leeming [Simplicity Media Ltd] wrote: > Hello, > > Sometimes it's difficult to get the attention of the core devs, they > have to prioritize like everyone else. > > One of the approaches used on the Django mailing list was a 5 for 1 > offer.. this is where you help out on 5 tickets (usually low priority > ones), and a senior/core dev will look at one of your tickets in return > (no guarantees of a fix, just purely to get their attention/comments) > > Would this approach be welcomed by the FS core team? Could be a great > way of not only getting more tickets looked at, whilst providing a > reward for doing so (as an alternative to bountys) > > Cal > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From cal.leeming at simplicitymedialtd.co.uk Wed Jan 2 21:11:33 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 2 Jan 2013 18:11:33 +0000 Subject: [Freeswitch-users] 5 for 1 offer - senior/core devs In-Reply-To: <50E47485.2010107@quentustech.com> References: <50E47485.2010107@quentustech.com> Message-ID: They use a really simplistic approach, no management of a karma system needed. http://www.mail-archive.com/django-developers at googlegroups.com/msg35324.html Like this: ------ I've triaged:https://code.djangoproject.com/ticket/15338https://code.djangoproject.com/ticket/16630https://code.djangoproject.com/ticket/18413https://code.djangoproject.com/ticket/12836https://code.djangoproject.com/ticket/18575 I was hoping someone could have a look at:*https://code.djangoproject.com/ticket/14567* I've also triaged:https://code.djangoproject.com/ticket/7377https://code.djangoproject.com/ticket/18682https://code.djangoproject.com/ticket/18761https://code.djangoproject.com/ticket/18742https://code.djangoproject.com/ticket/18829 I was hoping someone could have a look at: *https://code.djangoproject.com/ticket/11027* ---- On Wed, Jan 2, 2013 at 5:55 PM, William King wrote: > It would require someone who was willing to 'manage' the karma system. > Any links to how the django mailing list handles it? > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 01/02/2013 09:30 AM, Cal Leeming [Simplicity Media Ltd] wrote: > > Hello, > > > > Sometimes it's difficult to get the attention of the core devs, they > > have to prioritize like everyone else. > > > > One of the approaches used on the Django mailing list was a 5 for 1 > > offer.. this is where you help out on 5 tickets (usually low priority > > ones), and a senior/core dev will look at one of your tickets in return > > (no guarantees of a fix, just purely to get their attention/comments) > > > > Would this approach be welcomed by the FS core team? Could be a great > > way of not only getting more tickets looked at, whilst providing a > > reward for doing so (as an alternative to bountys) > > > > Cal > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/1c66fc51/attachment.html From schoch+freeswitch.org at xwin32.com Wed Jan 2 21:38:38 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 2 Jan 2013 10:38:38 -0800 Subject: [Freeswitch-users] You must define a domain called 'xx.com' in your directory In-Reply-To: References: Message-ID: On Fri, Dec 28, 2012 at 9:28 PM, Anton Kvashenkin wrote: > > > That's getting me closer! I have already changed the domain in vars.xml to include this line: This has caused '' to be expanded in both directory/default.xml and directory/xx.com.xml. I'm thinking the best way to avoid confusion is to either remove directory/default.xml, or remove directory/xx.com.xml, and use the default file, with the expanded domain name. On Sat, Dec 29, 2012 at 1:50 AM, Gabriel Gunderson wrote: > > Play around with find_user_xml and user_exists to trouble shoot your > directory independent of any specific SUA: > I have this user: freeswitch at internal> list_users user 103 userid|context|domain|group|contact|callgroup|effective_caller_id_name|effective_caller_id_number 103|default|xx.com|xx.com|error/user_not_registered|techsupport|Extension 103|103 But I still get this: 2013-01-02 10:21:33.913853 [WARNING] sofia_reg.c:2485 Can't find user [ 103 at xx.com] You must define a domain called 'xx.com' in your directory and add a user with the id="103" attribute and you must configure your device to use the proper domain in it's authentication credentials. and... freeswitch at internal> user_exists id 103 xx.com false -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/f395fba7/attachment-0001.html From william.king at quentustech.com Wed Jan 2 21:51:54 2013 From: william.king at quentustech.com (William King) Date: Wed, 02 Jan 2013 10:51:54 -0800 Subject: [Freeswitch-users] 5 for 1 offer - senior/core devs In-Reply-To: References: <50E47485.2010107@quentustech.com> Message-ID: <50E481CA.3000500@quentustech.com> Seems like an interesting system. I would be concerned about it just increasing the level of noise. If I saw an email like that on the mailing list I would be much more likely to take a look. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 01/02/2013 10:11 AM, Cal Leeming [Simplicity Media Ltd] wrote: > They use a really simplistic approach, no management of a karma system > needed. > > http://www.mail-archive.com/django-developers at googlegroups.com/msg35324.html > > Like this: > > ------ > > I've triaged: > https://code.djangoproject.com/ticket/15338 > https://code.djangoproject.com/ticket/16630 > https://code.djangoproject.com/ticket/18413 > https://code.djangoproject.com/ticket/12836 > https://code.djangoproject.com/ticket/18575 > > I was hoping someone could have a look at:* > https://code.djangoproject.com/ticket/14567* > > I've also triaged: > https://code.djangoproject.com/ticket/7377 > https://code.djangoproject.com/ticket/18682 > https://code.djangoproject.com/ticket/18761 > https://code.djangoproject.com/ticket/18742 > https://code.djangoproject.com/ticket/18829 > > I was hoping someone could have a look at: > *https://code.djangoproject.com/ticket/11027* > > ---- > > On Wed, Jan 2, 2013 at 5:55 PM, William King > > wrote: > > It would require someone who was willing to 'manage' the karma system. > Any links to how the django mailing list handles it? > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 01/02/2013 09:30 AM, Cal Leeming [Simplicity Media Ltd] wrote: > > Hello, > > > > Sometimes it's difficult to get the attention of the core devs, they > > have to prioritize like everyone else. > > > > One of the approaches used on the Django mailing list was a 5 for 1 > > offer.. this is where you help out on 5 tickets (usually low priority > > ones), and a senior/core dev will look at one of your tickets in > return > > (no guarantees of a fix, just purely to get their attention/comments) > > > > Would this approach be welcomed by the FS core team? Could be a great > > way of not only getting more tickets looked at, whilst providing a > > reward for doing so (as an alternative to bountys) > > > > Cal > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From william.king at quentustech.com Wed Jan 2 21:56:52 2013 From: william.king at quentustech.com (William King) Date: Wed, 02 Jan 2013 10:56:52 -0800 Subject: [Freeswitch-users] Freeswitch TLS and Yealink t26p In-Reply-To: <1356090896.23186.170.camel@marces.madrid.commsmundi.com> References: <1356082510.23186.154.camel@marces.madrid.commsmundi.com> <1356090896.23186.170.camel@marces.madrid.commsmundi.com> Message-ID: <50E482F4.8060701@quentustech.com> Another thing to be aware of is that there is an outstanding bug where Yealinks are not able to load certain custom CA's because the time on the phone is not being synced to NTP before the CA is validated and loaded. Once you changed the transport have you run into any new issues? William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 12/21/2012 03:54 AM, Antonio wrote: > Answer to myself.... > > In the yealink configuration, in the account parameters, the "transport" > must be force to TLS. > > I don't know why it just works.... Before i was using DNS-SRV, that > should be the first option, yealink should have some issue here... i > will report to them. > > > Thanks, > Ant?nio > > On Fri, 2012-12-21 at 10:35 +0100, Antonio wrote: >> Hi, >> >> I'm trying to register a yealink with TLS, using my one certificates. >> >> I follow the wiki and In fs i have both agent.pem and cafile.pem . I >> install in the phone the root certificate. >> >> But when i try to register, i have (tport log): >> >> >> tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x808fb0) msg >> 0x7fe9d0aa8180 from (udp/192.168.10.1:5060) has 340 bytes, veclen = 1 >> tport.c:3004 tport_deliver() tport_deliver(0x808fb0): msg >> 0x7fe9d0aa8180 (340 bytes) from udp/192.168.10.23:5060/sip next=(nil) >> tport.c:4202 tport_release() tport_release(0x808fb0): 0x7fe9d01142f0 >> by 0x7fe9d025d920 with 0x7fe9d0aa8180 >> tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fe9c802aad0): >> events IN >> tport.c:869 tport_alloc_secondary() >> tport_alloc_secondary(0x7fe9c802aad0): new secondary tport 0x7fe9c03e8450 >> tport_type_tls.c:603 tport_tls_accept() >> tport_tls_accept(0x7fe9c03e8450): new connection from >> tls/192.168.10.36:48754/sips >> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events >> NEGOTIATING >> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events >> NEGOTIATING >> tport_tls.c:526 tls_post_connection_check() >> tls_post_connection_check(0x7fe9c03e8450): Peer did not provide X.509 >> Certificate. >> >> >> >> I could make it work and have a register in the tls profile when i >> check on the phone the option in Security->Trusted Certificates: "Only >> Accept Trusted Certificates: DISABLED". >> Could it be some bug in the yealink, or I?m missing something in the >> conf... >> >> Another question, is there any problem if i choose to use this >> configuration... since is the phone that ignores the certificate and >> the validation is done by the server and not by the client. >> >> Can you help me? >> >> Thanks, >> Ant?nio >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > > Un cordial saludo / Best regards, > > _________________________ > > Ant?nio Silva > > E-mail:asilva at wirelessmundi.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Jan 2 22:50:42 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Jan 2013 11:50:42 -0800 Subject: [Freeswitch-users] regarding attendent transfer In-Reply-To: <50DD1E02.7090503@gmail.com> References: <50DC447C.1080703@gmail.com> <50DD1E02.7090503@gmail.com> Message-ID: Okay, then if it isn't properly transferring the call you will need to look at the console debug log. You can put the output into pastebin.freeswitch.org. -MC On Thu, Dec 27, 2012 at 8:20 PM, veerabhadrarao` < bhadrarao.kankatala at gmail.com> wrote: > ** > On 12/27/2012 11:24 PM, Michael Collins wrote: > > Are you using the *4 + extension number to do an attended transfer? Also, > keep in mind that the *4 attended transfer will *only* transfer to a > registered user on the system, that is, it transfers to "user/xxxx" where > xxxx is the destination number that you dial after pressing *4. > > -MC > > On Thu, Dec 27, 2012 at 4:52 AM, veerabhadrarao` < > bhadrarao.kankatala at gmail.com> wrote: > > hai > > Can anyone plaese tell me how to make attendent transfer. I am able to > working on Blind transfer using default.xml and features.xml but not > attendenet transfer successfully. I am using Freeswitch 1.2.4 version. > > thank you > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > Hai thank you for your Response > > I am dialling *4 and Extension then the call is Establishing From > B-->C after that how can i connect A--->C. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/c76e1205/attachment.html From roger.castaldo at gmail.com Wed Jan 2 23:34:25 2013 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Wed, 2 Jan 2013 15:34:25 -0500 Subject: [Freeswitch-users] Configuration System Message-ID: Hi everyone, I have been working on a configuration gui for freeswitch for several years now. It has gone through several revisions and changes in its core and well as the interface itself, basically making it more efficient and more user friendly as well as more expandable. It was a decision as of late now to take that work and make it open source, it is kind of giving out a lot of work at no cost, but I came to a point with it that I just cannot do it myself anymore. It is written entirely in C# and is its own self contained web server. The code is available at http://code.google.com/p/freeswitchconfig/. I would like to get others on board to assist in testing, coding,etc. I was looking to add this to the wiki but do not have access and looking to port it over to have a branch that both has a self contained web server and one that will run inside an asp.net capable web server. Enjoy the usage of it for those that do end up using it, any and all feed back would be appreciated. Also on a side note it is a web based system that has its own mobile version that automatically switches to accordingly. I just moved it to google code so have had no time to work on the wiki or anything for it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130102/3f700d42/attachment-0001.html From dujinfang at gmail.com Thu Jan 3 02:46:43 2013 From: dujinfang at gmail.com (Seven Du) Date: Thu, 3 Jan 2013 07:46:43 +0800 Subject: [Freeswitch-users] mod_mp4 working and usage In-Reply-To: References: Message-ID: According to the code mod_mp4 only works with mp4 that has correct hint track. AFAIK Xlite 5.0 only support H263 but not H264, while mp4 container can contain mp3 video most mp4 has AAC audio and H264 video. We have some experiment code that can play video via mod_vlc, follow this thread http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/085558.html On Wednesday, January 2, 2013 at 7:05 PM, Santosh Kadam wrote: > Hello there, > > Can someone explain if the mod_mp4 is working ? If yes, can someone forward the procedure to make the same work ? > > Currently - I am using FreeSwitch with Xlite 5.0 as my SIP client. Does this client support playback of MP4 video? If not, please suggest an appropriate SIP client as well. > > thanks in advance for the response. > > Warm Regards > santosh > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130103/7634367d/attachment.html From gabe at gundy.org Thu Jan 3 03:41:51 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 2 Jan 2013 17:41:51 -0700 Subject: [Freeswitch-users] Configuration System In-Reply-To: References: Message-ID: On Wed, Jan 2, 2013 at 1:34 PM, Roger Castaldo wrote: > Hi everyone, I have been working on a configuration gui for freeswitch for > several years now. It has gone through several revisions and changes in its > core and well as the interface itself, basically making it more efficient > and more user friendly as well as more expandable. It was a decision as of > late now to take that work and make it open source, it is kind of giving out > a lot of work at no cost, but I came to a point with it that I just cannot > do it myself anymore. Thanks for contribution under the MPL! > I just moved it to google code so have had no time to work on the wiki or anything for it. Just a thought, you might get more take up with a git based repo. I notice google code is not nearly as active as it was years ago. Heck, you might even see about getting it into FreeSWITCH's contrib. Anyway, I'll pass it along to my friends that hack on FreeSWITCH and C#. Good luck! Gabe From jmesquita at freeswitch.org Thu Jan 3 07:21:49 2013 From: jmesquita at freeswitch.org (Jmesquita@freeswitch.org) Date: Thu, 3 Jan 2013 01:21:49 -0300 Subject: [Freeswitch-users] Naming convention - sip_profile_name / sofia_profile_name / sip_gateway_name In-Reply-To: References: Message-ID: <9FDA0269-7E7B-4770-AB2B-9FBDAEF3A891@freeswitch.org> Someone should give this guy the documentation award or something... Thank you for your contributions Cal! Regards, Jo?o Mesquita On 30/12/2012, at 04:12 p.m., "Cal Leeming [Simplicity Media Ltd]" wrote: > Hello, > > The following ticket has been raised in regards to the somewhat confusion naming convention of sip_profile_name / sofia_profile_name / sip_gateway_name. > > http://jira.freeswitch.org/browse/FS-4980 > > To avoid confusion, I have also updated the documentation to reflect the current behavior; > > http://wiki.freeswitch.org/wiki/Variable_sofia_profile_name > http://wiki.freeswitch.org/wiki/Variable_sip_profile_name > http://wiki.freeswitch.org/wiki/Variable_sip_gateway_name > > If anyone spots any mistakes in the above, please let me know. > > Thanks > > Cal > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130103/4dca179d/attachment.html From mbrancaleoni at voismart.it Thu Jan 3 11:36:14 2013 From: mbrancaleoni at voismart.it (Matteo) Date: Thu, 3 Jan 2013 09:36:14 +0100 (CET) Subject: [Freeswitch-users] Sofia fails to detect offline gateway In-Reply-To: Message-ID: <121325363.1273.1357202174811.JavaMail.root@voismart.it> Hi, looking to the ping-max and ping-min code into sofia.c there's some "counter" logic broken. I've submitted a patch here: http://jira.freeswitch.org/browse/FS-4985 which seems to work here. With the patch applied, ping-min means "how many successful pings we must have before declaring a gw up". The interval between ping-min and ping-max is the "safe area" where a gw is marked as UP. so if we have, for example, min 3 and max 6, if the gw is up and we move counter between 3,4,5,6 the gw will be up. if from 6 we loose 4 (so counter == 2) pings in a row, the gw will be declared down. please note that on sofia startup the gw is always started as UP, so it will be up even if ping-min is > 1 . the "right" way starts when the gw goes down. perhaps if a gw is configured as "pinged" is reasonable to start it as down, but this is a different patch :) regards, matteo From a.venugopan at mundio.com Thu Jan 3 14:43:42 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 3 Jan 2013 11:43:42 +0000 Subject: [Freeswitch-users] FAX in Freeswitch Message-ID: <592A9CF93E12394E8472A6CC66E66BF233E10A@Mail-Kilo.squay.com> Hi, Is it possible to send/receive fax using Freeswitch? If so can you please let me know what changes need to be made? Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130103/a341f100/attachment.html From juanito1982 at gmail.com Thu Jan 3 15:05:19 2013 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Thu, 3 Jan 2013 13:05:19 +0100 Subject: [Freeswitch-users] FAX in Freeswitch In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233E10A@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF233E10A@Mail-Kilo.squay.com> Message-ID: Yes. Look at mod_spandsp Regards 2013/1/3 Archana Venugopan > Hi,**** > > Is it possible to send/receive fax using Freeswitch? If so can you please > let me know what changes need to be made?**** > > ** ** > > Regards,**** > > Archana**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130103/7310a796/attachment-0001.html From miha at softnet.si Thu Jan 3 15:08:24 2013 From: miha at softnet.si (Miha) Date: Thu, 03 Jan 2013 13:08:24 +0100 Subject: [Freeswitch-users] FAX in Freeswitch In-Reply-To: <592A9CF93E12394E8472A6CC66E66BF233E10A@Mail-Kilo.squay.com> References: <592A9CF93E12394E8472A6CC66E66BF233E10A@Mail-Kilo.squay.com> Message-ID: <50E574B8.8010102@softnet.si> Dne 1/3/2013 12:43 PM, pis(e Archana Venugopan: > > Hi, > > Is it possible to send/receive fax using Freeswitch? If so can you > please let me know what changes need to be made? > > Regards, > > Archana > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org http://wiki.freeswitch.org/wiki/Mod_spandsp miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130103/5fefc3d6/attachment.html From odermann at googlemail.com Thu Jan 3 15:20:25 2013 From: odermann at googlemail.com (Dennis) Date: Thu, 3 Jan 2013 13:20:25 +0100 Subject: [Freeswitch-users] Need to recompile fs after ODBC version update? In-Reply-To: References: Message-ID: steve, thank your for your information! kind regards dennis 2013/1/2 Steven Ayre : > Unless there has been an ABI change, if it's using the shared library it > should only require a restart. > > -Steve > > > > > On 2 January 2013 14:10, Dennis wrote: >> >> hi, >> >> i just like to ask, if someone could tell me, if i have to recompile >> fs after updating the odbc-version on the server? >> >> thanks & kind regrads >> dennis >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From neilp at cs.stanford.edu Thu Jan 3 16:46:40 2013 From: neilp at cs.stanford.edu (Neil Patel) Date: Thu, 3 Jan 2013 19:16:40 +0530 Subject: [Freeswitch-users] implementing missed call without a race Message-ID: Hi All, I recently implemented "missed call" functionality into my lua app: an inbound call comes in, I check the DB if it should be a "missed call". If so, the call is terminated and the caller is called back through an outbound call. This is useful in places like India where people are charged for outbound calls only. Thanks to help from folks on FS IRC, I implemented like so: 1. script1.lua takes the inbound call 2. If missed call is to be made, invoke script2.lua (passing script1's uuid) through a bgapi call, and let script1.lua terminate 3. in script2.lua, uuid_kill the session from script1.lua, and make an outbound call to the original caller Two questions: 1. Are there any problems with this implementation from correctness and performance standpoints? 2. I am using this app over a TDM (PRI) connection. During very heavy load, there is a race condition: after killing the original session but before initiating the new outbound call in script2, the available channel could be grabbed. Is there any way to fix or mitigate this race? Thanks, Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130103/cea075d2/attachment.html From mickstevens at yahoo.com Thu Jan 3 17:36:38 2013 From: mickstevens at yahoo.com (Mick Stevens) Date: Thu, 3 Jan 2013 06:36:38 -0800 (PST) Subject: [Freeswitch-users] Happy New Year! In-Reply-To: References: Message-ID: <1357223798.55422.YahooMailNeo@web160805.mail.bf1.yahoo.com> Hi Brian, Count me in, anything I can do to assist (in return for, hopefully, the acquisition of knowledge)? Assuming you approve, what are next steps? Might be worth us having a chat, so you know my background & can utilise me accordingly? ? Rgds, Mick Tel/SMS. +44(0)7967 594432 Fax. +44(0)7053 452429 Email/IM. mickstevens at yahoo.com Skype: mick_stevens www.facebook.com/mickstevens www.twitter.com/mickstevens ________________________________ From: Brian West To: FreeSWITCH Users Help Sent: Wednesday, 2 January 2013, 16:27 Subject: [Freeswitch-users] Happy New Year! http://blog.smartbear.com/software-quality/bid/167051/14-Ways-to-Contribute-to-Open-Source-without-Being-a-Programming-Genius-or-a-Rock-Star I need some folks to step up and really help out with Jira... Any takers? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266? |? F: +1.918.420.9267? |? M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 ISN: 410*543 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130103/7f6d5ef0/attachment.html From freeswitch at peely.com Thu Jan 3 18:55:23 2013 From: freeswitch at peely.com (peely) Date: Thu, 3 Jan 2013 07:55:23 -0800 (PST) Subject: [Freeswitch-users] Kudos to the FreSWITCH team! In-Reply-To: References: Message-ID: <1357228523207-7586031.post@n2.nabble.com> This has a whole lot more to do with interconnect agreements than how many CAPS one solution can send versus another. Frankly any old piece of SIP gear has the potential to send more CAPS than your average provider would expect to receive. Many telcos, especially tier one's will establish an upper limit on the number of attempts per second that any interconnect can initiate, for example BT in the UK limit it to one CAPS per 30 channels approximately. It's not fair to allow one customer to use the whole available capacity at the expense of other customers. We're forever clamping customers because their rubbish dialler software has no understanding of a 503 with Retry-After response and have gone to lengths to automate this to protect our network, as even FS can get crippled by a device which relentlessly initiates call after call (in some cases we've seen well over 100 CAPS from a single device). So when we detect a killer trunk we'll raise a critical event in the FS logs and use Fail2Ban to ignore requests from the customer's IP. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Kudos-to-the-FreSWITCH-team-tp7585975p7586031.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Thu Jan 3 19:09:59 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 03 Jan 2013 10:09:59 -0600 Subject: [Freeswitch-users] Kudos to the FreSWITCH team! In-Reply-To: <1357228523207-7586031.post@n2.nabble.com> Message-ID: 503 w/ Retry-After lol... Dialers ignore that one? Really? They would never do such a thing... The problem with Retry-after is how its implemented across the board... Not everyone interprets it the same way... Sofia for instance by default if it gets a 503 Retry After will HOLD that call for retry-after timeout, then send it again to the same destination, it does not however prevent new calls from going to the same gateway... (I found this little tid bit by accident trying to figure out why certain calls were not route advancing... And this is not something the FS developers did, but it was down deep in sofia) K On 1/3/13 9:55 AM, "peely" wrote: > > This has a whole lot more to do with interconnect agreements than how many > CAPS one solution can send versus another. Frankly any old piece of SIP gear > has the potential to send more CAPS than your average provider would expect > to receive. > > Many telcos, especially tier one's will establish an upper limit on the > number of attempts per second that any interconnect can initiate, for > example BT in the UK limit it to one CAPS per 30 channels approximately. > It's not fair to allow one customer to use the whole available capacity at > the expense of other customers. > > We're forever clamping customers because their rubbish dialler software has > no understanding of a 503 with Retry-After response and have gone to lengths > to automate this to protect our network, as even FS can get crippled by a > device which relentlessly initiates call after call (in some cases we've > seen well over 100 CAPS from a single device). So when we detect a killer > trunk we'll raise a critical event in the FS logs and use Fail2Ban to ignore > requests from the customer's IP. > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Kudos-to-the-FreSWITCH-team-tp75 > 85975p7586031.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From godson.g at gmail.com Thu Jan 3 19:11:49 2013 From: godson.g at gmail.com (Godson Gera) Date: Thu, 3 Jan 2013 21:41:49 +0530 Subject: [Freeswitch-users] Kudos to the FreSWITCH team! In-Reply-To: <1357228523207-7586031.post@n2.nabble.com> References: <1357228523207-7586031.post@n2.nabble.com> Message-ID: On Thu, Jan 3, 2013 at 9:25 PM, peely wrote: > > This has a whole lot more to do with interconnect agreements than how many > CAPS one solution can send versus another. This is true, most of the time its the CPS limit agreement with their upstream that forces providers to slow down the traffic way below than what any softswitch can blast out. -- Godson Gera IPPBX Solutions -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130103/6328fc15/attachment.html From freeswitch at peely.com Thu Jan 3 19:12:30 2013 From: freeswitch at peely.com (peely) Date: Thu, 3 Jan 2013 08:12:30 -0800 (PST) Subject: [Freeswitch-users] NAT traversal - the final say..! In-Reply-To: References: Message-ID: <1357229550070-7586032.post@n2.nabble.com> IMO as a provider, FS does an excellent job at detecting NAT on its own and handling it. We've rarely had any issue with our end sending INVITEs to the wrong place or one / no way speech. connectile dysfunction does nothing more. The only switch we enable is one to enable the SIP options ping and to use sip-force-expires to reduce the time the endpoint re-registers for mobile devices. As with others on this thread, the ALG in most routers is the culprit, I've rarely experienced one which does anything beneficial, most screw up the headers. We also recommend that customers change the SIP port on their handsets to anything other than 5060, older BT routers especially seem to stop forwarding responses after a few minutes, again as they have a very basic ALG. Using port forwarding in the router is the only sure fire way of getting reliable SIP through to a handset, Draytek routers seem to be pretty flaky until you do this or use TCP / TLS. If the router stops forwarding packets because of a poor ALG implementation or other firewall rules, nothing FS can do will overcome this. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/NAT-traversal-the-final-say-tp7585562p7586032.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at peely.com Thu Jan 3 19:29:00 2013 From: freeswitch at peely.com (peely) Date: Thu, 3 Jan 2013 08:29:00 -0800 (PST) Subject: [Freeswitch-users] Kudos to the FreSWITCH team! In-Reply-To: References: <1357228523207-7586031.post@n2.nabble.com> Message-ID: <1357230540266-7586035.post@n2.nabble.com> "A client (proxy or UAC) receiving a 503 (Service Unavailable) SHOULD attempt to forward the request to an alternate server. It SHOULD NOT forward any other requests to that server for the duration specified in the Retry-After header field, if present." But hey, if everyone stuck to the spec' then things would be really dull. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Kudos-to-the-FreSWITCH-team-tp7585975p7586035.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Thu Jan 3 19:38:29 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 03 Jan 2013 10:38:29 -0600 Subject: [Freeswitch-users] Kudos to the FreSWITCH team! In-Reply-To: <1357230540266-7586035.post@n2.nabble.com> Message-ID: I didn't say it was correct application on the spec, just pointing out how things actually work in real life... And for the commenter who said most soft switches can handle way more volume then they are allowed to, that's only true in certain situations... Look at some of the gear out there that's supposed to be good gear... They cant handle more then 25 to 50 calls per second... Not to mention over sell ratios on the end offices... How common is it to see 30+ to 1 oversells? On 1/3/13 10:29 AM, "peely" wrote: > "A client (proxy or UAC) receiving a 503 (Service Unavailable) SHOULD > attempt to forward the request to an alternate server. It SHOULD NOT > forward any other requests to that server for the duration specified > in the Retry-After header field, if present." > > But hey, if everyone stuck to the spec' then things would be really dull. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Kudos-to-the-FreSWITCH-team-tp75 > 85975p7586035.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From abaci64 at gmail.com Thu Jan 3 19:43:39 2013 From: abaci64 at gmail.com (Abaci) Date: Thu, 03 Jan 2013 11:43:39 -0500 Subject: [Freeswitch-users] Problem with Polycom In-Reply-To: <50E1BFC8.9060503@gmail.com> References: <50E1BFC8.9060503@gmail.com> Message-ID: <50E5B53B.6010903@gmail.com> Thinking of just allowing all traffic from OpenSIPS via ACL so that FreeSWITCH doesn't challenge the call again, my question is if there is a way to set the user (based on the FROM) on trusted calls (acl) before it hits the dialplan? I know you can do a set_user in the dialplan and then transfer the call but would like to avoid it. On 12/31/2012 11:39 AM, Abaci wrote: > We have OpenSIPS in front of FreeSWITCH, phones register to OpenSIPS, > OpenSIPS sends all calls to FreeSWITCH. the way we have it set up is > that both OpenSIPS and FreeSWITCH auth all the invites (we have it > like this for a reason). this setup works fine (with Yealink at least). > > Problem we have is with Polycom phones, the phone registers just fine > but can't make calls. When OpenSIPS gets the INVITE it responds with a > 407, to which the Polycom responds, OpenSIPS is then sending the call > to FreeSWITCH which sends a 407 again, OpenSIPS proxies it back to the > polycom, however the Polycom does not respond to the challenge. > > I don't think this is a FreeSWITCH problem, I would still like to hear > if anyone has an idea how I can get this working (config option?) > > Thanks From david.villasmil.work at gmail.com Thu Jan 3 20:00:05 2013 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 3 Jan 2013 18:00:05 +0100 Subject: [Freeswitch-users] Kudos to the FreSWITCH team! In-Reply-To: References: <1357230540266-7586035.post@n2.nabble.com> Message-ID: Hello guys, Thanks for the commets, but like I said: "The SWITCH is coming down"... Anyway, Thanks to FS-Team! David On Thu, Jan 3, 2013 at 5:38 PM, Ken Rice wrote: > I didn't say it was correct application on the spec, just pointing out how > things actually work in real life... > > And for the commenter who said most soft switches can handle way more > volume > then they are allowed to, that's only true in certain situations... Look at > some of the gear out there that's supposed to be good gear... They cant > handle more then 25 to 50 calls per second... Not to mention over sell > ratios on the end offices... How common is it to see 30+ to 1 oversells? > > > On 1/3/13 10:29 AM, "peely" wrote: > > > "A client (proxy or UAC) receiving a 503 (Service Unavailable) SHOULD > > attempt to forward the request to an alternate server. It SHOULD NOT > > forward any other requests to that server for the duration specified > > in the Retry-After header field, if present." > > > > But hey, if everyone stuck to the spec' then things would be really dull. > > > > > > > > -- > > View this message in context: > > > http://freeswitch-users.2379917.n2.nabble.com/Kudos-to-the-FreSWITCH-team-tp75 > > 85975p7586035.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130103/9610823f/attachment.html From gabe at gundy.org Thu Jan 3 20:04:00 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 3 Jan 2013 10:04:00 -0700 Subject: [Freeswitch-users] Happy New Year! In-Reply-To: <1357223798.55422.YahooMailNeo@web160805.mail.bf1.yahoo.com> References: <1357223798.55422.YahooMailNeo@web160805.mail.bf1.yahoo.com> Message-ID: On Thu, Jan 3, 2013 at 7:36 AM, Mick Stevens wrote: > Count me in, anything I can do to assist (in return for, hopefully, the > acquisition of knowledge)? I suspect that working in Jira is it's own reward in terms of knowledge acquisition. Dig in and let us know what you find. Best, Gabe From tahir at ictinnovations.com Thu Jan 3 20:25:44 2013 From: tahir at ictinnovations.com (Tahir Almas) Date: Thu, 3 Jan 2013 22:25:44 +0500 Subject: [Freeswitch-users] ICTFAX Version 2.1 released In-Reply-To: References: Message-ID: ICTDialer Version 1.0.0, A Freeswitch based autodialer application, released today with following enhancements * Improved Trunk management support * Select Trunk in campaign form * Select custom caller ID in campaign form * Set Max call duration in campaign form * Bug fix in campaign scheduling * Added Agent/Extenions management support * Added support for IVR Call transfer Application * Real-time Campaign Report updates * Sample import contact CSV file at GUI Fore more information, Please visit http://www.ictdialer.org Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130103/69990885/attachment-0001.html From madan.mallikarjun at outlook.com Thu Jan 3 22:04:34 2013 From: madan.mallikarjun at outlook.com (madan.mallikarjun) Date: Thu, 3 Jan 2013 11:04:34 -0800 (PST) Subject: [Freeswitch-users] Voice Mail Message-ID: <1357239874635-7586041.post@n2.nabble.com> hello all, is there a way where we can restrict or limit the number of voice mails per extensions created in F.S. regards Madan -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Voice-Mail-tp7586041.html Sent from the freeswitch-users mailing list archive at Nabble.com. From abaci64 at gmail.com Thu Jan 3 22:12:48 2013 From: abaci64 at gmail.com (Abaci) Date: Thu, 03 Jan 2013 14:12:48 -0500 Subject: [Freeswitch-users] Voice Mail In-Reply-To: <1357239874635-7586041.post@n2.nabble.com> References: <1357239874635-7586041.post@n2.nabble.com> Message-ID: <50E5D830.7070807@gmail.com> There is the "vm-disk-quota" param http://wiki.freeswitch.org/wiki/Mod_voicemail#vm-disk-quota I don't think there is something to limit the number of voicemails, if that's what you're looking for you can open a feature request on Jira. On 1/3/2013 2:04 PM, madan.mallikarjun wrote: > hello all, > > is there a way where we can restrict or limit the number of voice > mails per extensions created in F.S. > > regards > Madan > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Voice-Mail-tp7586041.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From royce3 at gmail.com Fri Jan 4 03:08:37 2013 From: royce3 at gmail.com (Royce Mitchell III) Date: Thu, 3 Jan 2013 18:08:37 -0600 Subject: [Freeswitch-users] mod_event_multicast on windows Message-ID: I'm trying to get multicast registrations working, and I'm stuck. Both servers are Window Server 2008 R2 64-bit FreeSWITCH executable is 32-bit I've disabled the Windows Firewall Both only have a single LAN adapter each. One has 3 ip addresses in different subnets ( 2 are for testing and can be removed if they are causing a problem ), the other server only has 1 ip address. I'm trying to replicate sofia registrations as described here: http://wiki.freeswitch.org/wiki/Mod_event_multicast#Replicating_Sofia_Registrations Here's my event_multicast.conf.xml: and here's what I get in the FreeSWITCH console on either machine: > load mod_event_multicast 2013-01-03 18:00:24.672251 [INFO] mod_enum.c:775 ENUM Reloaded 2013-01-03 18:00:24.672251 [INFO] switch_time.c:1028 Timezone reloaded 530 defin itions 2013-01-03 18:00:24.692252 [ERR] mod_event_multicast.c:421 Bind Error +OK Reloading XML -ERR [module load file routine returned an error] 2013-01-03 18:00:24.692252 [NOTICE] switch_event.c:450 Subclass reservation mult icast::event inuse by listeners, detaching.. 2013-01-03 18:00:24.692252 [CRIT] switch_loadable_module.c:929 Error Loading mod ule C:\FreeSWITCH\mod\mod_event_multicast.dll **Module load routine returned an error** I think it's likely I must be doing something wrong, but I can't seem to figure it out. I tried changing the multicast address to something random like 228.67.53.09, but that didn't help. If it's a problem with Windows' support of multicasting ( some googling seems to indicate winsock is weird about multicasting ), I'm happy to test changes, recompile and submit a patch to Jira, but I have no experience with multicast sockets, so I wanted to find out if it works for other people before digging into the code. -- Royce Mitchell IT Consultant ITAS Solutions royce3 at itas-solutions.com From gamar at center.com Fri Jan 4 01:42:33 2013 From: gamar at center.com (Gilbert Amar) Date: Thu, 3 Jan 2013 14:42:33 -0800 Subject: [Freeswitch-users] mod_event_socket event with no eventname Message-ID: <002401cdea03$9ee6cde0$dcb469a0$@center.com> Hello, I am using mod_event_socket org.freeswitch.esl.client-0.9.2 and FreeSWITCH Version 1.2.5.3. My app is build upon the Event Socket Outbound with : Socket-Mode: async Control: full event plain ALL I notice that I am receiving event like this Content-Length: 143 Content-Type: text/event-plain Command: sendmsg%20978b97b2-55a4-11e2-b38d-1f8230fa05b2 call-command: execute execute-app-name: playback execute-app-arg: phrase%3AAC_0002%3A Please notice the missing Event-Name The context : on an incoming call after playing some sound file I originate a new call and play a phrase to the outbound call. Not only this seems wrong to me but I receive this on the two connection that are involve. I attach the Wireshark pcap file. Gilbert -------------- next part -------------- A non-text attachment was scrubbed... Name: command.pcap.zip Type: application/x-zip-compressed Size: 19879 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130103/e100e602/attachment-0001.bin From brian at freeswitch.org Fri Jan 4 17:46:42 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Jan 2013 08:46:42 -0600 Subject: [Freeswitch-users] Happy New Year! In-Reply-To: <1357223798.55422.YahooMailNeo@web160805.mail.bf1.yahoo.com> References: <1357223798.55422.YahooMailNeo@web160805.mail.bf1.yahoo.com> Message-ID: Work jira, test bugs make sure the reports are correct. Help narrow down the issues if possible. /b On Jan 3, 2013, at 8:36 AM, Mick Stevens wrote: > Hi Brian, > > Count me in, anything I can do to assist (in return for, hopefully, the acquisition of knowledge)? > > Assuming you approve, what are next steps? Might be worth us having a chat, so you know my background & can utilise me accordingly? > > Rgds, Mick > Tel/SMS. +44(0)7967 594432 > Fax. +44(0)7053 452429 > > Email/IM. mickstevens at yahoo.com > Skype: mick_stevens > www.facebook.com/mickstevens > www.twitter.com/mickstevens > > > > ________________________________ > From: Brian West > To: FreeSWITCH Users Help > Sent: Wednesday, 2 January 2013, 16:27 > Subject: [Freeswitch-users] Happy New Year! > > http://blog.smartbear.com/software-quality/bid/167051/14-Ways-to-Contribute-to-Open-Source-without-Being-a-Programming-Genius-or-a-Rock-Star > > I need some folks to step up and really help out with Jira... Any takers? > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://freeswitchcookbook.com http://freeswitchbook.com T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130104/65b23fd7/attachment.html From brian at freeswitch.org Fri Jan 4 17:47:02 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Jan 2013 08:47:02 -0600 Subject: [Freeswitch-users] Happy New Year! In-Reply-To: References: <1357223798.55422.YahooMailNeo@web160805.mail.bf1.yahoo.com> Message-ID: <5D85F846-B9C3-41DC-9A7A-2E92F75793CD@freeswitch.org> You are correct you will learn a lot by doing that task. On Jan 3, 2013, at 11:04 AM, Gabriel Gunderson wrote: > On Thu, Jan 3, 2013 at 7:36 AM, Mick Stevens wrote: >> Count me in, anything I can do to assist (in return for, hopefully, the >> acquisition of knowledge)? > > I suspect that working in Jira is it's own reward in terms of > knowledge acquisition. > > Dig in and let us know what you find. > > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://freeswitchcookbook.com http://freeswitchbook.com T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130104/75b1556b/attachment.html From jmesquita at freeswitch.org Fri Jan 4 18:14:44 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 4 Jan 2013 12:14:44 -0300 Subject: [Freeswitch-users] Happy New Year! In-Reply-To: <5D85F846-B9C3-41DC-9A7A-2E92F75793CD@freeswitch.org> References: <1357223798.55422.YahooMailNeo@web160805.mail.bf1.yahoo.com> <5D85F846-B9C3-41DC-9A7A-2E92F75793CD@freeswitch.org> Message-ID: Actions normally are better than words! I already started going over the tickets! Happy hacking everyone! Jo?o Mesquita FreeSWITCH? Solutions On Fri, Jan 4, 2013 at 11:47 AM, Brian West wrote: > You are correct you will learn a lot by doing that task. > > On Jan 3, 2013, at 11:04 AM, Gabriel Gunderson wrote: > > On Thu, Jan 3, 2013 at 7:36 AM, Mick Stevens > wrote: > > Count me in, anything I can do to assist (in return for, hopefully, the > > acquisition of knowledge)? > > > I suspect that working in Jira is it's own reward in terms of > knowledge acquisition. > > Dig in and let us know what you find. > > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > http://freeswitchcookbook.com > http://freeswitchbook.com > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130104/7f56aa26/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Fri Jan 4 19:45:03 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 4 Jan 2013 16:45:03 +0000 Subject: [Freeswitch-users] Happy New Year! In-Reply-To: References: Message-ID: Hi Brian, Would you be happy to go along with the 5 for 1 approach? (see other email) Dealing with 5 tickets that have no relevance to my own work, and in return getting good attention on a ticket that does have relevance to my own work, would be killer :) I think this would really help encourage others to help out as well, gives people an incentive! Cal On Wed, Jan 2, 2013 at 4:27 PM, Brian West wrote: > > http://blog.smartbear.com/software-quality/bid/167051/14-Ways-to-Contribute-to-Open-Source-without-Being-a-Programming-Genius-or-a-Rock-Star > > I need some folks to step up and really help out with Jira... Any takers? > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > ISN: 410*543 > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130104/8026347b/attachment.html From kevin.jzh at hotmail.com Fri Jan 4 15:55:55 2013 From: kevin.jzh at hotmail.com (JinKevin) Date: Fri, 4 Jan 2013 20:55:55 +0800 Subject: [Freeswitch-users] Luasql.postgres loading issue Message-ID: Hi All, I'm using lua to the access the postgresql, but get the error as below. Any help is appreciated.I did a lot search in the list and on internet, however didn't find the solution. 2013-01-03 18:16:02.165155 [NOTICE] switch_ivr.c:1801 Transfer sofia/internal/kevin at 192.168.1.200 to XML[9908 at default] 2013-01-03 18:16:02.165155 [INFO] mod_dialplan_xml.c:557 Processing kevin ->9908 in context default 2013-01-03 18:16:02.243548 [ERR] mod_lua.cpp:198 error loading module 'luasql.postgres' from file '/usr/local/lib/lua/5.1/luasql/postgres.so': /usr/lib/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator stack traceback: [C]: ? [C]: in function 'require' /opt/freeswitch/scripts/cbdemo.lua:1: in main chunk Thanks,Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130104/76020ee9/attachment.html From victor.chukalovskiy at gmail.com Fri Jan 4 20:38:47 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Fri, 04 Jan 2013 12:38:47 -0500 Subject: [Freeswitch-users] Channel variables for npdi and rn on LRN dipped calls. Message-ID: <50E713A7.8080006@gmail.com> Hello and Happy New year! A SIP call comes in that was previously dipped. I'd like to be able to have "rn" and "npdi" available as channel variables that I can use in regex expressions. Is there a way to configure FreeSWITCH to put "rn" and "npdi" into separate channel variable? So far, info application shows me the following channel variables that contain LRN: Caller-Destination-Number: [5551234567;npdi=yes;rn=5555550001] variable_sip_req_user: [5551234567;npdi=yes;rn=5555550001] variable_sip_req_uri: [5551234567;npdi=yes;rn=5555550001 at 1.2.3.4:5060] I can probably dissect those and put "rn" and "npdi" into separate variables. But this means extra steps in dial-plan. In case there is a built-in way to accomplish the same task, it would be great! Could you please let me know. Thank you, Victor From msc at freeswitch.org Fri Jan 4 21:01:38 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Jan 2013 10:01:38 -0800 Subject: [Freeswitch-users] Configuration System In-Reply-To: References: Message-ID: If you haven't already done so please request a wiki account. I fell behind on approving new requests but now that the holidaze have passed I am all caught up. :) -MC On Wed, Jan 2, 2013 at 12:34 PM, Roger Castaldo wrote: > Hi everyone, I have been working on a configuration gui for freeswitch for > several years now. It has gone through several revisions and changes in > its core and well as the interface itself, basically making it more > efficient and more user friendly as well as more expandable. It was a > decision as of late now to take that work and make it open source, it is > kind of giving out a lot of work at no cost, but I came to a point with it > that I just cannot do it myself anymore. It is written entirely in C# and > is its own self contained web server. The code is available at > http://code.google.com/p/freeswitchconfig/. I would like to get others > on board to assist in testing, coding,etc. I was looking to add this to > the wiki but do not have access and looking to port it over to have a > branch that both has a self contained web server and one that will run > inside an asp.net capable web server. Enjoy the usage of it for those > that do end up using it, any and all feed back would be appreciated. Also > on a side note it is a web based system that has its own mobile version > that automatically switches to accordingly. I just moved it to google code > so have had no time to work on the wiki or anything for it. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130104/b29e43c6/attachment.html From jeff at jefflenk.com Fri Jan 4 22:11:57 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 4 Jan 2013 11:11:57 -0800 (PST) Subject: [Freeswitch-users] mod_event_multicast on windows In-Reply-To: References: Message-ID: <1357326717512-7586052.post@n2.nabble.com> I don't used that module but it should work. Are you binding to an IP Address that exists? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-event-multicast-on-windows-tp7586043p7586052.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Fri Jan 4 22:54:26 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 04 Jan 2013 13:54:26 -0600 Subject: [Freeswitch-users] Friday Free For All Activate! Message-ID: Hey Guys Friday Free For All! On the bridge! Get in here! sip:888 at conference.freeswitch.org Topic? Who knows its a Free For All! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130104/73ba5112/attachment.html From jeff at jefflenk.com Sat Jan 5 01:44:53 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 4 Jan 2013 14:44:53 -0800 (PST) Subject: [Freeswitch-users] Configuration System In-Reply-To: References: Message-ID: <1357339493054-7586054.post@n2.nabble.com> Roger good to hear from you! Thanks for pointing out your project(I had forgotten that you mentioned that you were working on this a while back). I would love to see a simple overview of what you are doing with it. I would also like to help maintaining any docs I could on the FS Wiki. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Configuration-System-tp7586020p7586054.html Sent from the freeswitch-users mailing list archive at Nabble.com. From sertys at gmail.com Sat Jan 5 02:25:44 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Sat, 5 Jan 2013 00:25:44 +0100 Subject: [Freeswitch-users] Freeswitch mod_sms and SIMPLE Message-ID: I stumbled upon this feature lately and since my clients support it, decided to add it. Wrote a lua wrapper over a web sms gateway and it's working correctly. I however found peer-to-peer chat very unreliable. There is no spooling of messages whenever one leg is unavailable and it isn't always delivering messages( i think this might be a client issue though). Who is maintaining the mod and what are the planned feature updates to it. I can write an external spooler, but i can bet there's a better event based approach in FS core. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130105/03bbca52/attachment.html From Hector.Geraldino at ipsoft.com Sat Jan 5 03:28:36 2013 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Sat, 5 Jan 2013 00:28:36 +0000 Subject: [Freeswitch-users] mod_event_socket event with no eventname In-Reply-To: <002401cdea03$9ee6cde0$dcb469a0$@center.com> References: <002401cdea03$9ee6cde0$dcb469a0$@center.com> Message-ID: You must filter the events on each channel, so you don't receive events on the outbound thread that are being generated on the inbound call thread. To accomplish this you have several options: use myevents on the outbound socket, or add a filter based on the call uuid. For the outbound call, you can have something like: final org.freeswitch.esl.client.inbound.Client client = new org.freeswitch.esl.client.inbound.Client(); client.connect("localhost", 8021, "ClueCon", 2); ... client.addEventFilter("Unique-ID", uuid); For the inbound call, in the class that extends AbstractOutboundClientHandler you can have something like: @Override protected void handleConnectResponse(ChannelHandlerContext ctx, EslEvent event) { String uuid = event.getEventHeaders().get(""Unique-ID"); sendSyncSingleLineCommand(channel, "filter Unique-ID " + uuid); ... } -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gilbert Amar Sent: Thursday, January 03, 2013 5:43 PM To: 'FreeSWITCH Users Help' Subject: [Freeswitch-users] mod_event_socket event with no eventname Hello, I am using mod_event_socket org.freeswitch.esl.client-0.9.2 and FreeSWITCH Version 1.2.5.3. My app is build upon the Event Socket Outbound with : Socket-Mode: async Control: full event plain ALL I notice that I am receiving event like this Content-Length: 143 Content-Type: text/event-plain Command: sendmsg%20978b97b2-55a4-11e2-b38d-1f8230fa05b2 call-command: execute execute-app-name: playback execute-app-arg: phrase%3AAC_0002%3A Please notice the missing Event-Name The context : on an incoming call after playing some sound file I originate a new call and play a phrase to the outbound call. Not only this seems wrong to me but I receive this on the two connection that are involve. I attach the Wireshark pcap file. Gilbert From dujinfang at gmail.com Sat Jan 5 05:38:13 2013 From: dujinfang at gmail.com (Seven Du) Date: Sat, 5 Jan 2013 10:38:13 +0800 Subject: [Freeswitch-users] mod_cdr_pg_csv on CentOS6.3 problem Message-ID: Hi, We just tried 1.2-stable and it compiles ok but failed to load with the following error libnssutil3.so: undefined symbol: PR_GetDirectorySeparator Looks there was problems on this and google shows most links related to freeswitch. And jira issues on this was to disable this with --without-pgsql. so my question is - Is there anyone using mod_cdr_pg_csv on CentOS 6.3? Or even the core pgsql ? https://www.google.com.hk/search?q=libnssutil3.so+PR_GetDirectorySeparator&rlz=1C5CHFA_enCN506CN506&aq=f&oq=li&aqs=chrome.2.57j59l3j60l2.3979&sugexp=chrome,mod=4&sourceid=chrome&ie=UTF-8 Thanks. Seven. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130105/3b69cf8f/attachment.html From kevin.jzh at hotmail.com Sat Jan 5 06:09:51 2013 From: kevin.jzh at hotmail.com (JinKevin) Date: Sat, 5 Jan 2013 11:09:51 +0800 Subject: [Freeswitch-users] Luasql.postgres loading issue In-Reply-To: References: Message-ID: Like to add some additional info:I'm using the latest stable freeswitch and Postgres9.2.1, runing on CentOS6.2.and built the luaSql per the wiki, the config file was changed as below T= postgres #T= sqlite#LUA_INC= $(PREFIX)/include LUA_INC= /usr/local/src/freeswitch/src/mod/languages/mod_lua/lua/ ######## PostgreSQL DRIVER_LIBS= -L/opt/postgres9.2/lib -lpq DRIVER_INCS= -I/opt/postgres9.2/include/.... Thanks,KevinFrom: kevin.jzh at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Fri, 4 Jan 2013 20:55:55 +0800 Subject: [Freeswitch-users] Luasql.postgres loading issue Hi All, I'm using lua to the access the postgresql, but get the error as below. Any help is appreciated. I did a lot search in the list and on internet, however didn't find the solution. 2013-01-03 18:16:02.165155 [NOTICE] switch_ivr.c:1801 Transfer sofia/internal/kevin at 192.168.1.200 to XML[9908 at default] 2013-01-03 18:16:02.165155 [INFO] mod_dialplan_xml.c:557 Processing kevin ->9908 in context default 2013-01-03 18:16:02.243548 [ERR] mod_lua.cpp:198 error loading module 'luasql.postgres' from file '/usr/local/lib/lua/5.1/luasql/postgres.so': /usr/lib/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator stack traceback: [C]: ? [C]: in function 'require' /opt/freeswitch/scripts/cbdemo.lua:1: in main chunk Thanks, Kevin _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130105/06a62591/attachment.html From fvillarroel at yahoo.com Sat Jan 5 19:06:56 2013 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Sat, 5 Jan 2013 08:06:56 -0800 (PST) Subject: [Freeswitch-users] Install ASTPP In-Reply-To: <50893659.7090001@gmail.com> Message-ID: <1357402016.8729.YahooMailClassic@web162003.mail.bf1.yahoo.com> http://wiki.astpp.org/index.php?title=Installation_Guide --- On Thu, 10/25/12, Joao Cesar / Consultor TI wrote: > From: Joao Cesar / Consultor TI > Subject: Re: [Freeswitch-users] Install ASTPP > To: freeswitch-users at lists.freeswitch.org > Date: Thursday, October 25, 2012, 9:53 AM > > Em 25/10/2012 10:46, Michael Jerris escreveu: > > Try sending this email to the mailing list? > > > > On Oct 25, 2012, at 8:41 AM, Joao Cesar / Consultor TI > > wrote: > > > >> Anyone know how to install the ASTPP really, > because the tutorial site, have not the slightest condition > to do. > >> Thank you in advance. > >> > >> -- > >> Joao Cesar S Pereira > >> Consultor TI > >> ---------------------------------------- > >> (31)3044-2576 > >> (31)8626-6943 > >> ---------------------------------------- > >> Gentileza, se nao houver necessidade > >> n?o imprima. A natureza agradece. > >> ---------------------------------------- > >> > >> > > -- > Joao Cesar S Pereira > Consultor TI > ---------------------------------------- > (31)3044-2576 > (31)8626-6943 > ---------------------------------------- > Gentileza, se nao houver necessidade > n?o imprima. A natureza agradece. > ---------------------------------------- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lloyd.aloysius at gmail.com Sat Jan 5 22:54:56 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Sat, 5 Jan 2013 14:54:56 -0500 Subject: [Freeswitch-users] Happy New Year! In-Reply-To: References: Message-ID: Hi Brian: Count me in Thanks Lloyd * * On Fri, Jan 4, 2013 at 11:45 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hi Brian, > > Would you be happy to go along with the 5 for 1 approach? (see other email) > > Dealing with 5 tickets that have no relevance to my own work, and in > return getting good attention on a ticket that does have relevance to my > own work, would be killer :) > > I think this would really help encourage others to help out as well, gives > people an incentive! > > Cal > > > On Wed, Jan 2, 2013 at 4:27 PM, Brian West wrote: > >> >> http://blog.smartbear.com/software-quality/bid/167051/14-Ways-to-Contribute-to-Open-Source-without-Being-a-Programming-Genius-or-a-Rock-Star >> >> I need some folks to step up and really help out with Jira... Any takers? >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH_Wire >> T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9266 >> UK: +44 20 3298 4900 >> ISN: 410*543 >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130105/08a9375a/attachment-0001.html From gabe at gundy.org Sun Jan 6 12:04:11 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Sun, 6 Jan 2013 02:04:11 -0700 Subject: [Freeswitch-users] FTC offers $50,000 in contest to end robocalls Message-ID: Will someone please forward this valuable infomation to Lenny? http://money.cnn.com/2013/01/02/pf/ftc-robocalls/index.html Gabe From sertys at gmail.com Sun Jan 6 14:03:58 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Sun, 6 Jan 2013 12:03:58 +0100 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: References: Message-ID: This is old news and he should be aware. On Jan 6, 2013 11:07 AM, "Gabriel Gunderson" wrote: > Will someone please forward this valuable infomation to Lenny? > > http://money.cnn.com/2013/01/02/pf/ftc-robocalls/index.html > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/714e3241/attachment.html From bdfoster at endigotech.com Sun Jan 6 16:39:25 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 6 Jan 2013 08:39:25 -0500 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: References: Message-ID: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> I doubt he is aware, given that he doesn't live in the US of A! Besides, the article is only four days old. Sent from my iPhone On Jan 6, 2013, at 6:03 AM, Daniel Ivanov wrote: > This is old news and he should be aware. > > On Jan 6, 2013 11:07 AM, "Gabriel Gunderson" wrote: >> Will someone please forward this valuable infomation to Lenny? >> >> http://money.cnn.com/2013/01/02/pf/ftc-robocalls/index.html >> >> >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/cfe939a4/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sun Jan 6 17:10:31 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 6 Jan 2013 14:10:31 +0000 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> References: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> Message-ID: So, who is this "Lenny" character?? Cal On Sun, Jan 6, 2013 at 1:39 PM, Brian Foster wrote: > I doubt he is aware, given that he doesn't live in the US of A! Besides, > the article is only four days old. > > Sent from my iPhone > > On Jan 6, 2013, at 6:03 AM, Daniel Ivanov wrote: > > This is old news and he should be aware. > On Jan 6, 2013 11:07 AM, "Gabriel Gunderson" wrote: > >> Will someone please forward this valuable infomation to Lenny? >> >> http://money.cnn.com/2013/01/02/pf/ftc-robocalls/index.html >> >> >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/d26a2f64/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sun Jan 6 17:12:17 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 6 Jan 2013 14:12:17 +0000 Subject: [Freeswitch-users] Channel variables for npdi and rn on LRN dipped calls. In-Reply-To: <50E713A7.8080006@gmail.com> References: <50E713A7.8080006@gmail.com> Message-ID: What the.. I thought "Caller-Destination-Number" was always the cleaned dialed number (i.e. 5551234567) - is this not the case? Cal On Fri, Jan 4, 2013 at 5:38 PM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Hello and Happy New year! > > A SIP call comes in that was previously dipped. I'd like to be able to > have "rn" and "npdi" available as channel variables that I can use in > regex expressions. Is there a way to configure FreeSWITCH to put "rn" > and "npdi" into separate channel variable? > > So far, info application shows me the following channel variables that > contain LRN: > > Caller-Destination-Number: [5551234567;npdi=yes;rn=5555550001] > variable_sip_req_user: [5551234567;npdi=yes;rn=5555550001] > variable_sip_req_uri: [5551234567;npdi=yes;rn=5555550001 at 1.2.3.4:5060] > > I can probably dissect those and put "rn" and "npdi" into separate > variables. But this means extra steps in dial-plan. In case there is a > built-in way to accomplish the same task, it would be great! > > Could you please let me know. > > Thank you, > Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/56bb103e/attachment-0001.html From darcy at Vex.Net Sun Jan 6 17:23:46 2013 From: darcy at Vex.Net (D'Arcy J.M. Cain) Date: Sun, 6 Jan 2013 09:23:46 -0500 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> References: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> Message-ID: <20130106092346.1c5fc1ac@dilbert> On Sun, 6 Jan 2013 08:39:25 -0500 Brian Foster wrote: > I doubt he is aware, given that he doesn't live in the US of A! > Besides, the article is only four days old. Maybe someone should call him and try to explain it. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net From ben at langfeld.co.uk Sun Jan 6 17:29:17 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Sun, 6 Jan 2013 14:29:17 +0000 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: <20130106092346.1c5fc1ac@dilbert> References: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> <20130106092346.1c5fc1ac@dilbert> Message-ID: This news was reported in October :) Regards, Ben Langfeld On 6 January 2013 14:23, D'Arcy J.M. Cain wrote: > On Sun, 6 Jan 2013 08:39:25 -0500 > Brian Foster wrote: > > I doubt he is aware, given that he doesn't live in the US of A! > > Besides, the article is only four days old. > > Maybe someone should call him and try to explain it. > > -- > D'Arcy J.M. Cain > System Administrator, Vex.Net > http://www.Vex.Net/ IM:darcy at Vex.Net > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/829addb5/attachment.html From bdfoster at endigotech.com Sun Jan 6 18:16:03 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 6 Jan 2013 10:16:03 -0500 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: References: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> <20130106092346.1c5fc1ac@dilbert> Message-ID: <0A29CF2E-01E6-4D4C-A58E-658DE4CD5034@endigotech.com> I'm pretty sure he's in the FS community. Sent from my iPhone On Jan 6, 2013, at 9:29 AM, Ben Langfeld wrote: > This news was reported in October :) > > Regards, > Ben Langfeld > > > On 6 January 2013 14:23, D'Arcy J.M. Cain wrote: >> On Sun, 6 Jan 2013 08:39:25 -0500 >> Brian Foster wrote: >> > I doubt he is aware, given that he doesn't live in the US of A! >> > Besides, the article is only four days old. >> >> Maybe someone should call him and try to explain it. >> >> -- >> D'Arcy J.M. Cain >> System Administrator, Vex.Net >> http://www.Vex.Net/ IM:darcy at Vex.Net >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/7d69f6c4/attachment.html From avi at avimarcus.net Sun Jan 6 20:00:52 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 6 Jan 2013 19:00:52 +0200 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: References: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> Message-ID: Cal: http://www.itslenny.com/ He says pre-recorded semi-intelligible things when there's silence on the line -- giving the illusion there's a (senile) person responding. Some of the recordings are quite hilarious! -Avi On Sun, Jan 6, 2013 at 4:10 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > So, who is this "Lenny" character?? > > Cal > > > On Sun, Jan 6, 2013 at 1:39 PM, Brian Foster wrote: > >> I doubt he is aware, given that he doesn't live in the US of A! Besides, >> the article is only four days old. >> >> Sent from my iPhone >> >> On Jan 6, 2013, at 6:03 AM, Daniel Ivanov wrote: >> >> This is old news and he should be aware. >> On Jan 6, 2013 11:07 AM, "Gabriel Gunderson" wrote: >> >>> Will someone please forward this valuable infomation to Lenny? >>> >>> http://money.cnn.com/2013/01/02/pf/ftc-robocalls/index.html >>> >>> >>> Gabe >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/82def773/attachment-0001.html From krice at freeswitch.org Sun Jan 6 20:19:52 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 06 Jan 2013 11:19:52 -0600 Subject: [Freeswitch-users] Channel variables for npdi and rn on LRN dipped calls. In-Reply-To: Message-ID: You have to parse out the NDPI and RN values... Just use a regex like ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6});ndpi=(*.);rn=([2-9]\d{2}[2-9]\d{6})$ That will get dialed digits in $1, ndpi value in $2, then LRN in $3, I would warn against trusting the LRN you are sent if you do not explicitly trust the customer sending you said LRN... Its not like its hard to spoof the LRN replacnig the LRN from a high cost ratecenter with a low cost ratecenter K On 1/6/13 8:12 AM, "Cal Leeming [Simplicity Media Ltd]" wrote: > What the.. > > I thought "Caller-Destination-Number" was always the cleaned dialed number > (i.e.?5551234567) - is this not the case? > > Cal > > On Fri, Jan 4, 2013 at 5:38 PM, Victor Chukalovskiy > wrote: >> Hello and Happy New year! >> >> A SIP call comes in that was previously dipped. I'd like to be able to >> have "rn" and "npdi" available as channel variables that I can use in >> regex expressions. Is there a way to configure FreeSWITCH to put "rn" >> and "npdi" into separate channel variable? >> >> So far, info application shows me the following channel variables that >> contain LRN: >> >> Caller-Destination-Number: [5551234567;npdi=yes;rn=5555550001] >> variable_sip_req_user: [5551234567;npdi=yes;rn=5555550001] >> variable_sip_req_uri: [5551234567;npdi=yes;rn=5555550001 at 1.2.3.4:5060 >> ] >> >> I can probably dissect those and put "rn" and "npdi" into separate >> variables. But this means extra steps in dial-plan. In case there is a >> built-in way to accomplish the same task, it would be great! >> >> Could you please let me know. >> >> Thank you, >> Victor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/1ac87d55/attachment.html From darcy at Vex.Net Sun Jan 6 20:54:10 2013 From: darcy at Vex.Net (D'Arcy J.M. Cain) Date: Sun, 6 Jan 2013 12:54:10 -0500 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: References: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> <20130106092346.1c5fc1ac@dilbert> Message-ID: <20130106125410.6629eb55@dilbert> On Sun, 6 Jan 2013 14:29:17 +0000 Ben Langfeld wrote: > This news was reported in October :) To Lenny? I want to hear that recording. :-) -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net From curriegrad2004 at gmail.com Sun Jan 6 21:10:38 2013 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 6 Jan 2013 10:10:38 -0800 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: <20130106125410.6629eb55@dilbert> References: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> <20130106092346.1c5fc1ac@dilbert> <20130106125410.6629eb55@dilbert> Message-ID: Dang Lenny's been reddited and now we have kidiots abusing the service... We need Ken to give us a new version of Lenny so people don't know it's him ;) On Sun, Jan 6, 2013 at 9:54 AM, D'Arcy J.M. Cain wrote: > On Sun, 6 Jan 2013 14:29:17 +0000 > Ben Langfeld wrote: >> This news was reported in October :) > > To Lenny? I want to hear that recording. :-) > > -- > D'Arcy J.M. Cain > System Administrator, Vex.Net > http://www.Vex.Net/ IM:darcy at Vex.Net > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From cal.leeming at simplicitymedialtd.co.uk Sun Jan 6 21:40:40 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 6 Jan 2013 18:40:40 +0000 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: References: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> Message-ID: Lmao - that's amusing. Cal On Sun, Jan 6, 2013 at 5:00 PM, Avi Marcus wrote: > Cal: > http://www.itslenny.com/ > He says pre-recorded semi-intelligible things when there's silence on the > line -- giving the illusion there's a (senile) person responding. > > Some of the recordings are quite hilarious! > -Avi > > > On Sun, Jan 6, 2013 at 4:10 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> So, who is this "Lenny" character?? >> >> Cal >> >> >> On Sun, Jan 6, 2013 at 1:39 PM, Brian Foster wrote: >> >>> I doubt he is aware, given that he doesn't live in the US of A! Besides, >>> the article is only four days old. >>> >>> Sent from my iPhone >>> >>> On Jan 6, 2013, at 6:03 AM, Daniel Ivanov wrote: >>> >>> This is old news and he should be aware. >>> On Jan 6, 2013 11:07 AM, "Gabriel Gunderson" wrote: >>> >>>> Will someone please forward this valuable infomation to Lenny? >>>> >>>> http://money.cnn.com/2013/01/02/pf/ftc-robocalls/index.html >>>> >>>> >>>> Gabe >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/aa02ba8d/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Sun Jan 6 23:01:24 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 6 Jan 2013 20:01:24 +0000 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension Message-ID: Could a core dev speculate on how much the expected bounty would be for this? Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/fbc6a501/attachment.html From avi at avimarcus.net Sun Jan 6 23:09:17 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 6 Jan 2013 22:09:17 +0200 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: References: Message-ID: You can basically already do this with ngrep, see: http://wiki.freeswitch.org/wiki/Packet_Capture#ngrep e.g.: ngrep -q '8005551212' -W byline port 5060 show packets on 5060 (might need to change port to 5080) to 8005551212. What would be far more awesome is an option for FS to attach the sip trace, or even the RTP, to the CDR, that way all debug info is together. -Avi Marcus On Sun, Jan 6, 2013 at 10:01 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Could a core dev speculate on how much the expected bounty would be for > this? > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/a1f3918b/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sun Jan 6 23:21:33 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 6 Jan 2013 20:21:33 +0000 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: References: Message-ID: That's a pretty good idea - ty :) I agree having all this info together would be pretty amazing, and is somewhat close to what I'm trying to achieve at the moment. Would it be a complex thing to modify FS to attach the sip trace to a CDR callback? I.e. are we talking a couple of thousand bucks in bounty? Cal On Sun, Jan 6, 2013 at 8:09 PM, Avi Marcus wrote: > You can basically already do this with ngrep, see: > http://wiki.freeswitch.org/wiki/Packet_Capture#ngrep > > e.g.: ngrep -q '8005551212' -W byline port 5060 > show packets on 5060 (might need to change port to 5080) to 8005551212. > > What would be far more awesome is an option for FS to attach the sip > trace, or even the RTP, to the CDR, that way all debug info is together. > > -Avi Marcus > > On Sun, Jan 6, 2013 at 10:01 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Could a core dev speculate on how much the expected bounty would be for >> this? >> >> Thanks >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/d2882083/attachment.html From jason.holden at start.ca Sun Jan 6 23:47:26 2013 From: jason.holden at start.ca (Jason Holden) Date: Sun, 6 Jan 2013 15:47:26 -0500 Subject: [Freeswitch-users] personal web portal for users to access voicemail and modify their extension settings Message-ID: Does anyone have any recommendations for this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/02bd1e5c/attachment.html From avi at avimarcus.net Mon Jan 7 00:06:34 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 6 Jan 2013 23:06:34 +0200 Subject: [Freeswitch-users] personal web portal for users to access voicemail and modify their extension settings In-Reply-To: References: Message-ID: We try to keep a list on: http://wiki.freeswitch.org/wiki/Freeswitch_Gui but there seem to be a few community projects not on there. FusionPBX has a community around it (and an active dev) Blue.box seems great, but I *think *the devs (a team) are more focused on cluster performance that individual PBX, but you'll have to judge for yourself. -Avi On Sun, Jan 6, 2013 at 10:47 PM, Jason Holden wrote: > Does anyone have any recommendations for this?**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/62068400/attachment-0001.html From bdfoster at endigotech.com Mon Jan 7 00:08:00 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 6 Jan 2013 16:08:00 -0500 Subject: [Freeswitch-users] personal web portal for users to access voicemail and modify their extension settings In-Reply-To: References: Message-ID: <51DF34FC-1FC5-4403-A941-5A252118BB42@endigotech.com> http://fusionpbx.com Sent from my iPhone On Jan 6, 2013, at 3:47 PM, "Jason Holden" wrote: > Does anyone have any recommendations for this? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130106/44b7da7a/attachment.html From dujinfang at gmail.com Mon Jan 7 09:41:12 2013 From: dujinfang at gmail.com (Seven Du) Date: Mon, 7 Jan 2013 14:41:12 +0800 Subject: [Freeswitch-users] unicode in CDR Message-ID: Hi We used Chinese UTF8 in chan vars and some times it's url encoded but sometimes in html entity, specifically in variables it's %xx but in caller_profile it's &#xxxx, all has the same "?142248". We have some difficulty on parsing this, any comment on this? %E4%BC%9A%E8%AE%AE142248 %E4%BC%9A%E8%AE%AE142248 %E4%BC%9A%E8%AE%AE142248 会议142248 会议142248 Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/0eadcf4c/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Jan 7 15:57:32 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 7 Jan 2013 12:57:32 +0000 Subject: [Freeswitch-users] unicode in CDR In-Reply-To: References: Message-ID: Hi Seven, Could you elaborate more on "having difficulty parsing this"? What are you parsing it with? What are you trying to achieve? What should be happening vs what is actually happening? Etc. Cal On Mon, Jan 7, 2013 at 6:41 AM, Seven Du wrote: > Hi > > We used Chinese UTF8 in chan vars and some times it's url encoded but > sometimes in html entity, specifically in variables it's %xx but in > caller_profile it's &#xxxx, all has the same "?142248". > > We have some difficulty on parsing this, any comment on this? > > > > > > %E4%BC%9A%E8%AE%AE142248 > > %E4%BC%9A%E8%AE%AE142248 > %E4%BC%9A%E8%AE%AE142248 > > > 会议142248 > 会议142248 > > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/dd289489/attachment.html From gvvsubhashkumar at gmail.com Mon Jan 7 16:01:35 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Mon, 7 Jan 2013 18:31:35 +0530 Subject: [Freeswitch-users] Log Rollover issue In-Reply-To: References: Message-ID: I used the latest binaries(Windows Installer),version info is given below and observered that when it is not able to rename the log file it is keep on writing the logs in freeswitch.log file which is crossing the limt set in logconf.xml file. The error seen in the freeswitch log file is [CRIT] mod_logfile.c:164 Error renaming log from C:/Program Files/FreeSWITCH/log/freeswitch.log.4 to C:/Program Files/FreeSWITCH/log/freeswitch.log.5 [No error] And the version info is Freswitch version 1.3.10 git b1b89e7 2012-12-14 22:53:48z Thanks, Subhash. On Thu, Nov 22, 2012 at 12:34 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Are you deleting the old ones? > It was probably already out of disk space when it was refusing to rename > the files. > > Looks like you have 40 of them, you may need to auto delete some. > > You are not providing much other data to go on so all I can do is guess. > The system call to move the file is failing which could be due to > permissions or disk full or whatever. > > If you want you can update to latest and I added a call to strerror(errno) > so you can see the exact reason. > > > > > On Tue, Nov 20, 2012 at 10:23 PM, Nageshwara Rao Moova < > mnrao2001 at gmail.com> wrote: > >> Hi Anthony, >> >> I doubt any permission issue as the failure happened not initially but >> after a while. The user we are using has sufficient permissions for doing >> rwx operations. Below is the log snippet from freeswitch log. >> >> If it?s any help, the freeswitch log had entries like this every few >> seconds: >> >> 2012-10-26 11:53:16.653473 [CRIT] mod_logfile.c:164 Error renaming log >> from /usr/local/freeswitch/log/freeswitch.log.40 to >> /usr/local/freeswitch/log/freeswitch.log.41 >> >> ?? >> >> 2012-10-27 00:19:30.313497 [CRIT] mod_logfile.c:164 Error renaming log >> from /usr/local/freeswitch/log/freeswitch.log.40 to >> /usr/local/freeswitch/log/freeswitch.log.41 >> >> >> Until it ran out of disk space and crashed. No reason why it couldn?t >> rename the file, though. >> >> >> >> >> On Tue, Nov 20, 2012 at 1:50 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> When it says it cannot, is that because of file permissions? Dos FS have >>> permission to write to the directory? >>> Can you reproduce this with logs and if so can you report it to Jira >>> http://jira.freeswitch.org >>> >>> >>> >>> On Mon, Nov 19, 2012 at 6:45 AM, Nageshwara Rao Moova < >>> mnrao2001 at gmail.com> wrote: >>> >>>> Hi all, >>>> >>>> I have modified my default logconf file for rollover to be restricted >>>> to 100. But sometimes freeswitch is unable to rename rollover say ?cannot >>>> rename freeswitch.log.41? and fails. But the fail follows with serious >>>> issue of writing all the logs to "freeswitch.log" and ends up filling the >>>> disk space. >>>> >>>> I have not change the default file size i.e 10MB. >>>> >>>> How does 10MB size overridden by freeswitch? >>>> >>>> -- >>>> regards & thanks >>>> -- >>>> m nageshwara rao >>>> 99891 86280 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> regards & thanks >> -- >> m nageshwara rao >> 99891 86280 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/ea3133a2/attachment-0001.html From avi at avimarcus.net Mon Jan 7 16:12:45 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 7 Jan 2013 15:12:45 +0200 Subject: [Freeswitch-users] SPA-2102 Gives 486 instead of Call Waiting Message-ID: Here's an intersting one that seems to have no results on google or in debug info: I have an SPA-2102 (firmware 5.2.10) that returns 486/Busy Here when a call comes in to the active line. "call waiting" is turned on, so it should be beeping in instead... I got these logs off the SPA but it doesn't seem very helpful: (and there should be more, I think) ==> /var/log/syslog <== Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]<<$FreeSWITCH: 5060(1148) Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]<<$FreeSWITCH: 5060(1148) Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]->$FreeSWITCH: 5060(312) Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]->$FreeSWITCH: 5060(312) Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il SIP/2.0 100 Trying Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il #012 Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il #012 Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]->$FreeSWITCH: 5060(338) Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il [0]->$FreeSWITCH: 5060(338) Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il SIP/2.0 486 Busy Here Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il #012 Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il #012 Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il [0]<<$FreeSWITCH: 5060(334) Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il [0]<<$FreeSWITCH: 5060(334) Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 I saw one suggestion to do a factory reset that fixed the problem, but that's a bit troublesome at the moment since the client can't load provisioning files (dumb network.) -Avi Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/88e258a3/attachment.html From dgarcia at anew.com.ve Mon Jan 7 17:17:10 2013 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Mon, 07 Jan 2013 09:47:10 -0430 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: References: Message-ID: <50EAD8E6.4050907@anew.com.ve> Hi Cal, Also I use this way to get a sip trace with filter criterias. Take a look to this link: http://www.jonathanmanning.com/2009/10/26/how-to-voip-sip-capture-with-tcpdump-on-linux/ With Wireshark you can apply additional filter On 1/6/2013 3:31 PM, Cal Leeming [Simplicity Media Ltd] wrote: > Could a core dev speculate on how much the expected bounty would be > for this? > > Thanks > > Cal > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2221 / Virus Database: 2637/5476 - Release Date: 12/21/12 > Internal Virus Database is out of date. > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/8bbaa572/attachment.html From dujinfang at gmail.com Mon Jan 7 18:19:43 2013 From: dujinfang at gmail.com (Seven Du) Date: Mon, 7 Jan 2013 23:19:43 +0800 Subject: [Freeswitch-users] unicode in CDR In-Reply-To: References: Message-ID: <860A09B3B6C24B85B8FC860669C0EAD1@gmail.com> I tried to parse that in Erlang and the xmerl XML lib seems has bug on parsing the unicode identity where I can only get one character "?" where it should be "??142248". But the real question is why it's different on different place in the XML with the same characters. Thanks. -- Seven Du Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, January 7, 2013 at 8:57 PM, Cal Leeming [Simplicity Media Ltd] wrote: > Hi Seven, > > Could you elaborate more on "having difficulty parsing this"? > > What are you parsing it with? > What are you trying to achieve? > What should be happening vs what is actually happening? > > Etc. > > Cal > > > On Mon, Jan 7, 2013 at 6:41 AM, Seven Du wrote: > > Hi > > > > We used Chinese UTF8 in chan vars and some times it's url encoded but sometimes in html entity, specifically in variables it's %xx but in caller_profile it's &#xxxx, all has the same "?142248". > > > > We have some difficulty on parsing this, any comment on this? > > > > > > > > > > %E4%BC%9A%E8%AE%AE142248 > > %E4%BC%9A%E8%AE%AE142248 > > %E4%BC%9A%E8%AE%AE142248 > > > > > > 会议142248 > > 会议142248 > > > > > > > > Thanks. > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/d4a7ccd2/attachment.html From steveayre at gmail.com Mon Jan 7 18:33:49 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 7 Jan 2013 15:33:49 +0000 Subject: [Freeswitch-users] unicode in CDR In-Reply-To: <860A09B3B6C24B85B8FC860669C0EAD1@gmail.com> References: <860A09B3B6C24B85B8FC860669C0EAD1@gmail.com> Message-ID: Looks like the caller_profile is dumped as a multibyte string while the variables are dumped as ascii (the individual bytes). I'd file a Jira report. On 7 January 2013 15:19, Seven Du wrote: > I tried to parse that in Erlang and the xmerl XML lib seems has bug on > parsing the unicode identity where I can only get one character "?" where > it should be "??142248". > > But the real question is why it's different on different place in the XML > with the same characters. > > Thanks. > > -- > Seven Du > Sent with Sparrow > > On Monday, January 7, 2013 at 8:57 PM, Cal Leeming [Simplicity Media Ltd] > wrote: > > Hi Seven, > > Could you elaborate more on "having difficulty parsing this"? > > What are you parsing it with? > What are you trying to achieve? > What should be happening vs what is actually happening? > > Etc. > > Cal > > > On Mon, Jan 7, 2013 at 6:41 AM, Seven Du wrote: > > Hi > > We used Chinese UTF8 in chan vars and some times it's url encoded but > sometimes in html entity, specifically in variables it's %xx but in > caller_profile it's &#xxxx, all has the same "?142248". > > We have some difficulty on parsing this, any comment on this? > > > > > > %E4%BC%9A%E8%AE%AE142248 > > %E4%BC%9A%E8%AE%AE142248 > %E4%BC%9A%E8%AE%AE142248 > > > 会议142248 > 会议142248 > > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/618a91b9/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Mon Jan 7 18:41:54 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 7 Jan 2013 15:41:54 +0000 Subject: [Freeswitch-users] unicode in CDR In-Reply-To: <860A09B3B6C24B85B8FC860669C0EAD1@gmail.com> References: <860A09B3B6C24B85B8FC860669C0EAD1@gmail.com> Message-ID: Ah, yeah dealing with unicode can be troublesome in many languages. >From my understanding - if FreeSWITCH has done something a certain way, then it will stay like that for the sake of backwards compatibility, despite whether it was the right/wrong way of doing something (unless it's an outright bug). You could push to have new variables defined with deprecation on the old, but these sorts of tickets are extremely low priority. Either way, this means that you probably won't see this being changed in the code (at least not for quite a long time). Generally speaking, the FreeSWITCH channel variables are not what a typical developer would consider 'sane'... i.e. you might need to clean/convert/verify etc. As Steven said, you can file a JIRA report, but unless the resulting data is absolutely wrong, then the chances of it being changed are slim. (If anyone thinks this is incorrect, please reply) Cal On Mon, Jan 7, 2013 at 3:19 PM, Seven Du wrote: > I tried to parse that in Erlang and the xmerl XML lib seems has bug on > parsing the unicode identity where I can only get one character "?" where > it should be "??142248". > > But the real question is why it's different on different place in the XML > with the same characters. > > Thanks. > > -- > Seven Du > Sent with Sparrow > > On Monday, January 7, 2013 at 8:57 PM, Cal Leeming [Simplicity Media Ltd] > wrote: > > Hi Seven, > > Could you elaborate more on "having difficulty parsing this"? > > What are you parsing it with? > What are you trying to achieve? > What should be happening vs what is actually happening? > > Etc. > > Cal > > > On Mon, Jan 7, 2013 at 6:41 AM, Seven Du wrote: > > Hi > > We used Chinese UTF8 in chan vars and some times it's url encoded but > sometimes in html entity, specifically in variables it's %xx but in > caller_profile it's &#xxxx, all has the same "?142248". > > We have some difficulty on parsing this, any comment on this? > > > > > > %E4%BC%9A%E8%AE%AE142248 > > %E4%BC%9A%E8%AE%AE142248 > %E4%BC%9A%E8%AE%AE142248 > > > 会议142248 > 会议142248 > > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/5c546224/attachment.html From marketing at cluecon.com Mon Jan 7 20:56:08 2013 From: marketing at cluecon.com (Michael Collins) Date: Mon, 7 Jan 2013 09:56:08 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Welcome to 2013! Things are slowly returning to normal now that we're past the holidays. We did manage to have community conference callsover the past few weeks. We spent most of our time going over some handy tips and tricks . We also discussed the new nested conditionsfeature. This week we will be having a community scrum and will be going over the status of the latest FreeSWITCH release. We will also be discussing a few ideas from the mailing list on how we might be able to get more volunteers working on Jiras. We would like to take a moment to say thanks to those who have made international DIDs available for the FreeSWITCH public conference. We now have DIDs for Australia, Germany, Israel, Spain, UK, and USA. The DIDs are listed on each conference call agenda page and there is a note as to who provided each DID. Thanks to all who have lent a hand. If you are in a position to provide a DID for other countries please contact me off list. Now that 2013 has hit we will be gearing up for the next ClueCon. There's only 210 days until ClueCon 2013! We will be sending out an official request for speaking proposals as well as contacting prospective event sponsors. We are looking forward to another great event. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/d2fef704/attachment.html From matt at inveroak.com Mon Jan 7 20:28:17 2013 From: matt at inveroak.com (Matt Broad) Date: Mon, 7 Jan 2013 17:28:17 +0000 Subject: [Freeswitch-users] Fwd: freeswitch on hold In-Reply-To: References: Message-ID: Hi I am wondering the best way to go about putting the below service together: Andy calls a number Andy's call is bridged to Brad Andy and Brad have conversation Brad then puts Andy on hold via DTMF Putting Andy on hold triggers a call to Carol When Carol answers she will be bridged with Brad whilst Andy is on hold Brad will then take Andy off hold via DTMF and all 3 will be connected Brad then hangsup and Andy and Carol are left talking until either hangsup -- Thanks Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/7251bba1/attachment.html From gavin.henry at gmail.com Mon Jan 7 22:33:49 2013 From: gavin.henry at gmail.com (Gavin Henry) Date: Mon, 7 Jan 2013 19:33:49 +0000 Subject: [Freeswitch-users] FS 1.2.5.3: Invalid Jitterbuffer spec [0] must be between 20 and 10000 Message-ID: Hi all, With 1.2.5.3 we're seeing a lot of this: sofia_glue.c:3549 Invalid Jitterbuffer spec [0] must be between 20 and 10000 our config hasn't changed. What can we read up on that has changed or is causing this? Thanks. From gabe at gundy.org Mon Jan 7 23:00:44 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 7 Jan 2013 13:00:44 -0700 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: References: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> <20130106092346.1c5fc1ac@dilbert> <20130106125410.6629eb55@dilbert> Message-ID: On Sun, Jan 6, 2013 at 11:10 AM, curriegrad2004 wrote: > Dang Lenny's been reddited and now we have kidiots abusing the > service... We need Ken to give us a new version of Lenny so people > don't know it's him ;) Do you have that link? Gabe From gabe at gundy.org Mon Jan 7 23:15:38 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 7 Jan 2013 13:15:38 -0700 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: References: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> <20130106092346.1c5fc1ac@dilbert> <20130106125410.6629eb55@dilbert> Message-ID: On Mon, Jan 7, 2013 at 1:00 PM, Gabriel Gunderson wrote: > Do you have that link? Never mind, I've started to figure out how to use reddit and have found the link(s). Yeah, I know, I'm the last person in the world to visit reddit. I work too much these days. Gabe From lloyd.aloysius at gmail.com Mon Jan 7 23:40:30 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Mon, 7 Jan 2013 15:40:30 -0500 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: <50EAD8E6.4050907@anew.com.ve> References: <50EAD8E6.4050907@anew.com.ve> Message-ID: I saw a demo from siproutes carrier. Their cdr have sip trace for every record and they keep for three months , then archive. It is awesome if freeswitch attach sip trace to the cdr. May be we can setup a bounty for this. * * On Mon, Jan 7, 2013 at 9:17 AM, Saugort Dario Garcia Tovar < dgarcia at anew.com.ve> wrote: > Hi Cal, > > Also I use this way to get a sip trace with filter criterias. Take a look > to this link: > > > http://www.jonathanmanning.com/2009/10/26/how-to-voip-sip-capture-with-tcpdump-on-linux/ > > With Wireshark you can apply additional filter > > > > On 1/6/2013 3:31 PM, Cal Leeming [Simplicity Media Ltd] wrote: > > Could a core dev speculate on how much the expected bounty would be for > this? > > Thanks > > Cal > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2221 / Virus Database: 2637/5476 - Release Date: 12/21/12 > Internal Virus Database is out of date. > > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/bf255daa/attachment.html From victor.chukalovskiy at gmail.com Mon Jan 7 23:45:33 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Mon, 07 Jan 2013 15:45:33 -0500 Subject: [Freeswitch-users] Channel variables for npdi and rn on LRN dipped calls. In-Reply-To: References: Message-ID: <50EB33ED.6010806@gmail.com> Ken, Thank you very much for feedback. I'll then follow your regex example to parse it. -Victor On 13-01-06 12:19 PM, Ken Rice wrote: > Re: [Freeswitch-users] Channel variables for npdi and rn on LRN dipped > calls. You have to parse out the NDPI and RN values... > > Just use a regex like > ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6});ndpi=(*.);rn=([2-9]\d{2}[2-9]\d{6})$ > That will get dialed digits in $1, ndpi value in $2, then LRN in $3, > > I would warn against trusting the LRN you are sent if you do not > explicitly trust the customer sending you said LRN... Its not like its > hard to spoof the LRN replacnig the LRN from a high cost ratecenter > with a low cost ratecenter > > K > > On 1/6/13 8:12 AM, "Cal Leeming [Simplicity Media Ltd]" > wrote: > > What the.. > > I thought "Caller-Destination-Number" was always the cleaned > dialed number (i.e. 5551234567) - is this not the case? > > Cal > > On Fri, Jan 4, 2013 at 5:38 PM, Victor Chukalovskiy > wrote: > > Hello and Happy New year! > > A SIP call comes in that was previously dipped. I'd like to be > able to > have "rn" and "npdi" available as channel variables that I can > use in > regex expressions. Is there a way to configure FreeSWITCH to > put "rn" > and "npdi" into separate channel variable? > > So far, info application shows me the following channel > variables that > contain LRN: > > Caller-Destination-Number: [5551234567;npdi=yes;rn=5555550001] > variable_sip_req_user: [5551234567;npdi=yes;rn=5555550001] > variable_sip_req_uri: > [5551234567;npdi=yes;rn=5555550001 at 1.2.3.4:5060 > ] > > I can probably dissect those and put "rn" and "npdi" into separate > variables. But this means extra steps in dial-plan. In case > there is a > built-in way to accomplish the same task, it would be great! > > Could you please let me know. > > Thank you, > Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/cfef07ca/attachment.html From bdfoster at endigotech.com Mon Jan 7 23:53:56 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 7 Jan 2013 15:53:56 -0500 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: References: <212F1653-2DE6-4ACD-83BF-1F4DAAC5703E@endigotech.com> <20130106092346.1c5fc1ac@dilbert> <20130106125410.6629eb55@dilbert> Message-ID: <4B1A5D45-86AF-412C-A7E3-EAD31141A8A9@endigotech.com> Hey you're farther along than I am in regards to reddit lol Sent from my iPhone On Jan 7, 2013, at 3:15 PM, Gabriel Gunderson wrote: > On Mon, Jan 7, 2013 at 1:00 PM, Gabriel Gunderson wrote: >> Do you have that link? > > Never mind, I've started to figure out how to use reddit and have > found the link(s). Yeah, I know, I'm the last person in the world to > visit reddit. I work too much these days. > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveayre at gmail.com Tue Jan 8 00:03:32 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 7 Jan 2013 21:03:32 +0000 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: References: <50EAD8E6.4050907@anew.com.ve> Message-ID: Current logging is the Sofia-SIP stack calling the sofia.c:logger() callback for the transport log. At this level packets are not attached to any call, so matching it to a call would require a 2nd parsing of the packet and matching it based on Call-ID... it would be quite a performance hit. The callback functions where mod_sofia handles a packet matched to a call receives the packet from the Sofia-SIP stack via the sip_t data structure. This contains the parsed message, not the raw packet. This makes it unsuitable for logging. I'm not sure if any other parameters passed to the callback contain the raw packet, which would be required to implement the feature. One of the mod_sofia developers would have a better grasp of the Sofia-SIP API though. -Steve On 7 January 2013 20:40, Lloyd Aloysius wrote: > I saw a demo from siproutes carrier. Their cdr have sip trace for every > record and they keep for three months , then archive. > > It is awesome if freeswitch attach sip trace to the cdr. May be we can setup > a bounty for this. > > > > > > On Mon, Jan 7, 2013 at 9:17 AM, Saugort Dario Garcia Tovar > wrote: >> >> Hi Cal, >> >> Also I use this way to get a sip trace with filter criterias. Take a look >> to this link: >> >> >> http://www.jonathanmanning.com/2009/10/26/how-to-voip-sip-capture-with-tcpdump-on-linux/ >> >> With Wireshark you can apply additional filter >> >> >> >> On 1/6/2013 3:31 PM, Cal Leeming [Simplicity Media Ltd] wrote: >> >> Could a core dev speculate on how much the expected bounty would be for >> this? >> >> Thanks >> >> Cal >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2221 / Virus Database: 2637/5476 - Release Date: 12/21/12 >> Internal Virus Database is out of date. >> >> >> >> -- >> Atentamente, >> Dario Garc?a >> Consultor. >> >> CCCT, Nivel C2, Sector Yarey, Mz, >> Ofc. MZ03a. >> Caracas-Venezuela. >> Tel?fono: +58 212 9081842 >> Cel: +58 412 2221515 >> dgarcia at anew.com.ve >> http://www.anew.com.ve >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nbhatti at gmail.com Tue Jan 8 00:12:03 2013 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 8 Jan 2013 00:12:03 +0300 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: References: <50EAD8E6.4050907@anew.com.ve> Message-ID: <246917E3-02B5-4FA1-B141-3B3E5DA8524E@gmail.com> http://www.sipcapture.org/ http://code.google.com/p/homer/wiki/HOWTO Thanks, -- Muhammad Naseer Bhatti On Jan 7, 2013, at 11:40 PM, Lloyd Aloysius wrote: > I saw a demo from siproutes carrier. Their cdr have sip trace for every record and they keep for three months , then archive. > > It is awesome if freeswitch attach sip trace to the cdr. May be we can setup a bounty for this. > > > > > > On Mon, Jan 7, 2013 at 9:17 AM, Saugort Dario Garcia Tovar wrote: > Hi Cal, > > Also I use this way to get a sip trace with filter criterias. Take a look to this link: > > http://www.jonathanmanning.com/2009/10/26/how-to-voip-sip-capture-with-tcpdump-on-linux/ > > With Wireshark you can apply additional filter > > > > On 1/6/2013 3:31 PM, Cal Leeming [Simplicity Media Ltd] wrote: >> Could a core dev speculate on how much the expected bounty would be for this? >> >> Thanks >> >> Cal >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2221 / Virus Database: 2637/5476 - Release Date: 12/21/12 >> Internal Virus Database is out of date. >> > > > -- > Atentamente, > Dario Garc?a > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/25293f55/attachment.html From gavin.henry at gmail.com Tue Jan 8 00:15:00 2013 From: gavin.henry at gmail.com (Gavin Henry) Date: Mon, 7 Jan 2013 21:15:00 +0000 Subject: [Freeswitch-users] FS 1.2.5.3: Invalid Jitterbuffer spec [0] must be between 20 and 10000 In-Reply-To: References: Message-ID: > Hi all, > > With 1.2.5.3 we're seeing a lot of this: > > sofia_glue.c:3549 Invalid Jitterbuffer spec [0] must be between 20 and 10000 > > our config hasn't changed. What can we read up on that has changed or > is causing this? We've found it. This is in sofia_glue.c if (jb_msec < 20 || jb_msec > 10000) { 3549 switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(tech_pvt->session), SWITCH_LOG_ERROR, 3550 "Invalid Jitterbuffer spec [%d] must be between 20 and 10000\n", jb_msec); 3551 } else { and mod_spandsp_fax is setting it: http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_spandsp/mod_spandsp_fax.c?r=a2e46d558f07366cc99bbf8d5967c3b438bbfd9a#to1347: switch_channel_set_variable(channel, "jitterbuffer_msec", "0") Every time we get a fax :-0 Will send a Jira -- http://www.suretecsystems.com/services/openldap/ http://www.surevoip.co.uk From anthony.minessale at gmail.com Tue Jan 8 00:16:57 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Jan 2013 15:16:57 -0600 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: References: <50EAD8E6.4050907@anew.com.ve> Message-ID: have you seen the homer project? On Mon, Jan 7, 2013 at 3:03 PM, Steven Ayre wrote: > Current logging is the Sofia-SIP stack calling the sofia.c:logger() > callback for the transport log. At this level packets are not attached > to any call, so matching it to a call would require a 2nd parsing of > the packet and matching it based on Call-ID... it would be quite a > performance hit. > > The callback functions where mod_sofia handles a packet matched to a > call receives the packet from the Sofia-SIP stack via the sip_t data > structure. This contains the parsed message, not the raw packet. This > makes it unsuitable for logging. I'm not sure if any other parameters > passed to the callback contain the raw packet, which would be required > to implement the feature. One of the mod_sofia developers would have a > better grasp of the Sofia-SIP API though. > > -Steve > > > > > > > On 7 January 2013 20:40, Lloyd Aloysius wrote: > > I saw a demo from siproutes carrier. Their cdr have sip trace for every > > record and they keep for three months , then archive. > > > > It is awesome if freeswitch attach sip trace to the cdr. May be we can > setup > > a bounty for this. > > > > > > > > > > > > On Mon, Jan 7, 2013 at 9:17 AM, Saugort Dario Garcia Tovar > > wrote: > >> > >> Hi Cal, > >> > >> Also I use this way to get a sip trace with filter criterias. Take a > look > >> to this link: > >> > >> > >> > http://www.jonathanmanning.com/2009/10/26/how-to-voip-sip-capture-with-tcpdump-on-linux/ > >> > >> With Wireshark you can apply additional filter > >> > >> > >> > >> On 1/6/2013 3:31 PM, Cal Leeming [Simplicity Media Ltd] wrote: > >> > >> Could a core dev speculate on how much the expected bounty would be for > >> this? > >> > >> Thanks > >> > >> Cal > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> No virus found in this message. > >> Checked by AVG - www.avg.com > >> Version: 2012.0.2221 / Virus Database: 2637/5476 - Release Date: > 12/21/12 > >> Internal Virus Database is out of date. > >> > >> > >> > >> -- > >> Atentamente, > >> Dario Garc?a > >> Consultor. > >> > >> CCCT, Nivel C2, Sector Yarey, Mz, > >> Ofc. MZ03a. > >> Caracas-Venezuela. > >> Tel?fono: +58 212 9081842 > >> Cel: +58 412 2221515 > >> dgarcia at anew.com.ve > >> http://www.anew.com.ve > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/4f1a1234/attachment-0001.html From gavin.henry at gmail.com Tue Jan 8 00:22:10 2013 From: gavin.henry at gmail.com (Gavin Henry) Date: Mon, 7 Jan 2013 21:22:10 +0000 Subject: [Freeswitch-users] FS 1.2.5.3: Invalid Jitterbuffer spec [0] must be between 20 and 10000 In-Reply-To: References: Message-ID: > Will send a Jira FYI - http://jira.freeswitch.org/browse/FS-5002 > -- > http://www.suretecsystems.com/services/openldap/ > http://www.surevoip.co.uk -- http://www.suretecsystems.com/services/openldap/ http://www.surevoip.co.uk From tomasz.szuster at gmail.com Tue Jan 8 00:40:08 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Mon, 7 Jan 2013 22:40:08 +0100 Subject: [Freeswitch-users] Fwd: freeswitch on hold In-Reply-To: References: Message-ID: Hi Matt, Some time ago I've the same questions. The answer is *4. After you press this you can bridge two separate calls Brad and Carol. If you have 3 people on the same line then maybe you have to create converence, don't know this. But start with *4 :) More experienced users will help you with this. Regards. Tom. On Mon, Jan 7, 2013 at 6:28 PM, Matt Broad wrote: > Hi > > I am wondering the best way to go about putting the below service together: > > Andy calls a number > Andy's call is bridged to Brad > Andy and Brad have conversation > Brad then puts Andy on hold via DTMF > Putting Andy on hold triggers a call to Carol > When Carol answers she will be bridged with Brad whilst Andy is on hold > Brad will then take Andy off hold via DTMF and all 3 will be connected > Brad then hangsup and Andy and Carol are left talking until either hangsup > > -- > Thanks Matt > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Pozdrawiam Tomasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/90a4b1f0/attachment.html From william.king at quentustech.com Tue Jan 8 00:47:33 2013 From: william.king at quentustech.com (William King) Date: Mon, 07 Jan 2013 13:47:33 -0800 Subject: [Freeswitch-users] Freeswitch mod_sms and SIMPLE In-Reply-To: References: Message-ID: <50EB4275.2080304@quentustech.com> What specifically do you mean by peer to peer? There are two methods for sending SMS messages: one that validates that the message was sent, and one that gives a single attempt. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 01/04/2013 03:25 PM, Daniel Ivanov wrote: > I stumbled upon this feature lately and since my clients support it, > decided to add it. Wrote a lua wrapper over a web sms gateway and it's > working correctly. I however found peer-to-peer chat very unreliable. > There is no spooling of messages whenever one leg is unavailable and it > isn't always delivering messages( i think this might be a client issue > though). Who is maintaining the mod and what are the planned feature > updates to it. I can write an external spooler, but i can bet there's a > better event based approach in FS core. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveayre at gmail.com Tue Jan 8 00:46:32 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 7 Jan 2013 21:46:32 +0000 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: References: <50EAD8E6.4050907@anew.com.ve> Message-ID: I have. Personally I use my own custom daemon that simply uses libpcap to capture and logs packets on port 5060/5080 to pcap files rotated on time/size limits which're stored for N days then deleted by a cronjob. I agree a separate project is probably a good way to log the SIP traffic... Cal, a project like Homer allows you to capture the SIP traffic from a separate server on the same LAN segment or using a monitor port of a managed switch. That offloads the SIP capture entirely away from the CPU/Disk of the SIP server, which can have performance benefits for your softswitch server. Additionally using a pcap file format can have other benefits, since it allows you to use existing tools like wireshark for analysis. Searching for a call in a pcap file is trivial using the Call-ID header which you'll have in the sip_call_id channel variable. That said, the idea of storing a history of the received SIP packets within the CDR itself is interesting and would have benefits on simpler installs such as a small office PBX - although I wouldn't necessarily do it by default. -Steve On 7 January 2013 21:16, Anthony Minessale wrote: > have you seen the homer project? > > > On Mon, Jan 7, 2013 at 3:03 PM, Steven Ayre wrote: >> >> Current logging is the Sofia-SIP stack calling the sofia.c:logger() >> callback for the transport log. At this level packets are not attached >> to any call, so matching it to a call would require a 2nd parsing of >> the packet and matching it based on Call-ID... it would be quite a >> performance hit. >> >> The callback functions where mod_sofia handles a packet matched to a >> call receives the packet from the Sofia-SIP stack via the sip_t data >> structure. This contains the parsed message, not the raw packet. This >> makes it unsuitable for logging. I'm not sure if any other parameters >> passed to the callback contain the raw packet, which would be required >> to implement the feature. One of the mod_sofia developers would have a >> better grasp of the Sofia-SIP API though. >> >> -Steve >> >> >> >> >> >> >> On 7 January 2013 20:40, Lloyd Aloysius wrote: >> > I saw a demo from siproutes carrier. Their cdr have sip trace for every >> > record and they keep for three months , then archive. >> > >> > It is awesome if freeswitch attach sip trace to the cdr. May be we can >> > setup >> > a bounty for this. >> > >> > >> > >> > >> > >> > On Mon, Jan 7, 2013 at 9:17 AM, Saugort Dario Garcia Tovar >> > wrote: >> >> >> >> Hi Cal, >> >> >> >> Also I use this way to get a sip trace with filter criterias. Take a >> >> look >> >> to this link: >> >> >> >> >> >> >> >> http://www.jonathanmanning.com/2009/10/26/how-to-voip-sip-capture-with-tcpdump-on-linux/ >> >> >> >> With Wireshark you can apply additional filter >> >> >> >> >> >> >> >> On 1/6/2013 3:31 PM, Cal Leeming [Simplicity Media Ltd] wrote: >> >> >> >> Could a core dev speculate on how much the expected bounty would be for >> >> this? >> >> >> >> Thanks >> >> >> >> Cal >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> No virus found in this message. >> >> Checked by AVG - www.avg.com >> >> Version: 2012.0.2221 / Virus Database: 2637/5476 - Release Date: >> >> 12/21/12 >> >> Internal Virus Database is out of date. >> >> >> >> >> >> >> >> -- >> >> Atentamente, >> >> Dario Garc?a >> >> Consultor. >> >> >> >> CCCT, Nivel C2, Sector Yarey, Mz, >> >> Ofc. MZ03a. >> >> Caracas-Venezuela. >> >> Tel?fono: +58 212 9081842 >> >> Cel: +58 412 2221515 >> >> dgarcia at anew.com.ve >> >> http://www.anew.com.ve >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From william.king at quentustech.com Tue Jan 8 00:48:27 2013 From: william.king at quentustech.com (William King) Date: Mon, 07 Jan 2013 13:48:27 -0800 Subject: [Freeswitch-users] Luasql.postgres loading issue In-Reply-To: References: Message-ID: <50EB42AB.6060407@quentustech.com> I would advise that you don't use the luasql method. Instead use the FS core pgsql support. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 01/04/2013 07:09 PM, JinKevin wrote: > > Like to add some additional info: > I'm using the latest stable freeswitch and Postgres9.2.1, runing on > CentOS6.2. > and built the luaSql per the wiki, the config file was changed as below > > T= postgres > #T= sqlite > #LUA_INC= $(PREFIX)/include > LUA_INC= /usr/local/src/freeswitch/src/mod/languages/mod_lua/lua/ > ######## PostgreSQL > DRIVER_LIBS= -L/opt/postgres9.2/lib -lpq > DRIVER_INCS= -I/opt/postgres9.2/include/ > .... > > Thanks, > Kevin > ------------------------------------------------------------------------ > From: kevin.jzh at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Fri, 4 Jan 2013 20:55:55 +0800 > Subject: [Freeswitch-users] Luasql.postgres loading issue > > Hi All, > > I'm using lua to the access the postgresql, but get the error as below. > Any help is appreciated. > I did a lot search in the list and on internet, however didn't find the > solution. > > 2013-01-03 18:16:02.165155 [NOTICE] switch_ivr.c:1801 Transfer > sofia/internal/kevin at 192.168.1.200 > to XML[9908 at default] > 2013-01-03 18:16:02.165155 [INFO] mod_dialplan_xml.c:557 Processing > kevin ->9908 in context default > 2013-01-03 18:16:02.243548 [ERR] mod_lua.cpp:198 error loading module > 'luasql.postgres' from file '/usr/local/lib/lua/5.1/luasql/postgres.so': > /usr/lib/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator > stack traceback: > [C]: ? > [C]: in function 'require' > /opt/freesw itch/scripts/cbdemo.lua:1: in main chunk > > > Thanks, > Kevin > > _________________________________________________________________________ Professional > FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The > CudaTel Communication Server Official FreeSWITCH > Sites http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From devel at omninet.eu Tue Jan 8 01:45:48 2013 From: devel at omninet.eu (Anestis Mavro) Date: Tue, 8 Jan 2013 00:45:48 +0200 Subject: [Freeswitch-users] stop unwanted subscriptions for presence of other users Message-ID: Hello, In an installation with several groups, where you don't want to enable users from one group to subscribe and get information about the other group, presence should be somehow protected from "unauthorized" subscriptions. Is there a way to achieve this on user level? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/f854068d/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Tue Jan 8 02:21:19 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 7 Jan 2013 23:21:19 +0000 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: References: <50EAD8E6.4050907@anew.com.ve> Message-ID: >From previous experience in web development, I know that sometimes the most obvious way isn't the right way. So, until I've tried it myself, I really can't say whether or not having the packets with the CDRs would be the right/wrong thing to do. However, HOMER on a switch port mirror looks very interesting - thank you for this recommendation. I'm going to be spending some time looking into all these different approaches, then I'll post back once I'm a bit more exposed to it. Really appreciate the feedback from everyone on this Cal On Mon, Jan 7, 2013 at 9:46 PM, Steven Ayre wrote: > I have. > > Personally I use my own custom daemon that simply uses libpcap to > capture and logs packets on port 5060/5080 to pcap files rotated on > time/size limits which're stored for N days then deleted by a cronjob. > > I agree a separate project is probably a good way to log the SIP traffic... > > Cal, a project like Homer allows you to capture the SIP traffic from a > separate server on the same LAN segment or using a monitor port of a > managed switch. That offloads the SIP capture entirely away from the > CPU/Disk of the SIP server, which can have performance benefits for > your softswitch server. Additionally using a pcap file format can have > other benefits, since it allows you to use existing tools like > wireshark for analysis. Searching for a call in a pcap file is trivial > using the Call-ID header which you'll have in the sip_call_id channel > variable. > > That said, the idea of storing a history of the received SIP packets > within the CDR itself is interesting and would have benefits on > simpler installs such as a small office PBX - although I wouldn't > necessarily do it by default. > > -Steve > > > > On 7 January 2013 21:16, Anthony Minessale > wrote: > > have you seen the homer project? > > > > > > On Mon, Jan 7, 2013 at 3:03 PM, Steven Ayre wrote: > >> > >> Current logging is the Sofia-SIP stack calling the sofia.c:logger() > >> callback for the transport log. At this level packets are not attached > >> to any call, so matching it to a call would require a 2nd parsing of > >> the packet and matching it based on Call-ID... it would be quite a > >> performance hit. > >> > >> The callback functions where mod_sofia handles a packet matched to a > >> call receives the packet from the Sofia-SIP stack via the sip_t data > >> structure. This contains the parsed message, not the raw packet. This > >> makes it unsuitable for logging. I'm not sure if any other parameters > >> passed to the callback contain the raw packet, which would be required > >> to implement the feature. One of the mod_sofia developers would have a > >> better grasp of the Sofia-SIP API though. > >> > >> -Steve > >> > >> > >> > >> > >> > >> > >> On 7 January 2013 20:40, Lloyd Aloysius > wrote: > >> > I saw a demo from siproutes carrier. Their cdr have sip trace for > every > >> > record and they keep for three months , then archive. > >> > > >> > It is awesome if freeswitch attach sip trace to the cdr. May be we can > >> > setup > >> > a bounty for this. > >> > > >> > > >> > > >> > > >> > > >> > On Mon, Jan 7, 2013 at 9:17 AM, Saugort Dario Garcia Tovar > >> > wrote: > >> >> > >> >> Hi Cal, > >> >> > >> >> Also I use this way to get a sip trace with filter criterias. Take a > >> >> look > >> >> to this link: > >> >> > >> >> > >> >> > >> >> > http://www.jonathanmanning.com/2009/10/26/how-to-voip-sip-capture-with-tcpdump-on-linux/ > >> >> > >> >> With Wireshark you can apply additional filter > >> >> > >> >> > >> >> > >> >> On 1/6/2013 3:31 PM, Cal Leeming [Simplicity Media Ltd] wrote: > >> >> > >> >> Could a core dev speculate on how much the expected bounty would be > for > >> >> this? > >> >> > >> >> Thanks > >> >> > >> >> Cal > >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> >> > >> >> > >> >> No virus found in this message. > >> >> Checked by AVG - www.avg.com > >> >> Version: 2012.0.2221 / Virus Database: 2637/5476 - Release Date: > >> >> 12/21/12 > >> >> Internal Virus Database is out of date. > >> >> > >> >> > >> >> > >> >> -- > >> >> Atentamente, > >> >> Dario Garc?a > >> >> Consultor. > >> >> > >> >> CCCT, Nivel C2, Sector Yarey, Mz, > >> >> Ofc. MZ03a. > >> >> Caracas-Venezuela. > >> >> Tel?fono: +58 212 9081842 > >> >> Cel: +58 412 2221515 > >> >> dgarcia at anew.com.ve > >> >> http://www.anew.com.ve > >> >> > >> >> > >> >> > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> > > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/add035cc/attachment.html From anthony.minessale at gmail.com Tue Jan 8 02:35:17 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 7 Jan 2013 17:35:17 -0600 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: References: <50EAD8E6.4050907@anew.com.ve> Message-ID: The best bet would probably to be using the call-id from the cdr to generate a link over to homer to look at a particular trace similar to how fisheye and jira work together. On Mon, Jan 7, 2013 at 5:21 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > >From previous experience in web development, I know that sometimes the > most obvious way isn't the right way. > > So, until I've tried it myself, I really can't say whether or not having > the packets with the CDRs would be the right/wrong thing to do. > > However, HOMER on a switch port mirror looks very interesting - thank you > for this recommendation. > > I'm going to be spending some time looking into all these different > approaches, then I'll post back once I'm a bit more exposed to it. > > Really appreciate the feedback from everyone on this > > Cal > > > On Mon, Jan 7, 2013 at 9:46 PM, Steven Ayre wrote: > >> I have. >> >> Personally I use my own custom daemon that simply uses libpcap to >> capture and logs packets on port 5060/5080 to pcap files rotated on >> time/size limits which're stored for N days then deleted by a cronjob. >> >> I agree a separate project is probably a good way to log the SIP >> traffic... >> >> Cal, a project like Homer allows you to capture the SIP traffic from a >> separate server on the same LAN segment or using a monitor port of a >> managed switch. That offloads the SIP capture entirely away from the >> CPU/Disk of the SIP server, which can have performance benefits for >> your softswitch server. Additionally using a pcap file format can have >> other benefits, since it allows you to use existing tools like >> wireshark for analysis. Searching for a call in a pcap file is trivial >> using the Call-ID header which you'll have in the sip_call_id channel >> variable. >> >> That said, the idea of storing a history of the received SIP packets >> within the CDR itself is interesting and would have benefits on >> simpler installs such as a small office PBX - although I wouldn't >> necessarily do it by default. >> >> -Steve >> >> >> >> On 7 January 2013 21:16, Anthony Minessale >> wrote: >> > have you seen the homer project? >> > >> > >> > On Mon, Jan 7, 2013 at 3:03 PM, Steven Ayre >> wrote: >> >> >> >> Current logging is the Sofia-SIP stack calling the sofia.c:logger() >> >> callback for the transport log. At this level packets are not attached >> >> to any call, so matching it to a call would require a 2nd parsing of >> >> the packet and matching it based on Call-ID... it would be quite a >> >> performance hit. >> >> >> >> The callback functions where mod_sofia handles a packet matched to a >> >> call receives the packet from the Sofia-SIP stack via the sip_t data >> >> structure. This contains the parsed message, not the raw packet. This >> >> makes it unsuitable for logging. I'm not sure if any other parameters >> >> passed to the callback contain the raw packet, which would be required >> >> to implement the feature. One of the mod_sofia developers would have a >> >> better grasp of the Sofia-SIP API though. >> >> >> >> -Steve >> >> >> >> >> >> >> >> >> >> >> >> >> >> On 7 January 2013 20:40, Lloyd Aloysius >> wrote: >> >> > I saw a demo from siproutes carrier. Their cdr have sip trace for >> every >> >> > record and they keep for three months , then archive. >> >> > >> >> > It is awesome if freeswitch attach sip trace to the cdr. May be we >> can >> >> > setup >> >> > a bounty for this. >> >> > >> >> > >> >> > >> >> > >> >> > >> >> > On Mon, Jan 7, 2013 at 9:17 AM, Saugort Dario Garcia Tovar >> >> > wrote: >> >> >> >> >> >> Hi Cal, >> >> >> >> >> >> Also I use this way to get a sip trace with filter criterias. Take a >> >> >> look >> >> >> to this link: >> >> >> >> >> >> >> >> >> >> >> >> >> http://www.jonathanmanning.com/2009/10/26/how-to-voip-sip-capture-with-tcpdump-on-linux/ >> >> >> >> >> >> With Wireshark you can apply additional filter >> >> >> >> >> >> >> >> >> >> >> >> On 1/6/2013 3:31 PM, Cal Leeming [Simplicity Media Ltd] wrote: >> >> >> >> >> >> Could a core dev speculate on how much the expected bounty would be >> for >> >> >> this? >> >> >> >> >> >> Thanks >> >> >> >> >> >> Cal >> >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> consulting at freeswitch.org >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org >> >> >> http://wiki.freeswitch.org >> >> >> http://www.cluecon.com >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> >> >> No virus found in this message. >> >> >> Checked by AVG - www.avg.com >> >> >> Version: 2012.0.2221 / Virus Database: 2637/5476 - Release Date: >> >> >> 12/21/12 >> >> >> Internal Virus Database is out of date. >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> Atentamente, >> >> >> Dario Garc?a >> >> >> Consultor. >> >> >> >> >> >> CCCT, Nivel C2, Sector Yarey, Mz, >> >> >> Ofc. MZ03a. >> >> >> Caracas-Venezuela. >> >> >> Tel?fono: +58 212 9081842 >> >> >> Cel: +58 412 2221515 >> >> >> dgarcia at anew.com.ve >> >> >> http://www.anew.com.ve >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> consulting at freeswitch.org >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org >> >> >> http://wiki.freeswitch.org >> >> >> http://www.cluecon.com >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> >> >> >> > >> >> > >> >> > >> >> > >> _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > googletalk:conf+888 at conference.freeswitch.org >> > pstn:+19193869900 >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/47dcacb9/attachment-0001.html From msc at freeswitch.org Tue Jan 8 03:48:51 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jan 2013 16:48:51 -0800 Subject: [Freeswitch-users] Fwd: freeswitch on hold In-Reply-To: References: Message-ID: Just as an FYI, the *4 is talking about the bind meta app that is found in the vanilla configs, Local_Extension, in conf/dialplan/default.xml. Keep in mind that this only works when the person pressing *4 is the recipient (i.e. the b-leg). Be sure to set up a separate extension if you want the a-leg to be able to dial *4. (Don't forget, the "default" dialplan is an example, so be sure to review it and make appropriate changes.) -MC On Mon, Jan 7, 2013 at 1:40 PM, Tomasz Szuster wrote: > Hi Matt, > > Some time ago I've the same questions. > The answer is *4. > After you press this you can bridge two separate calls Brad and Carol. > > If you have 3 people on the same line then maybe you have to create > converence, don't know this. > > But start with *4 :) > > More experienced users will help you with this. > > Regards. > Tom. > > On Mon, Jan 7, 2013 at 6:28 PM, Matt Broad wrote: > >> Hi >> >> I am wondering the best way to go about putting the below service >> together: >> >> Andy calls a number >> Andy's call is bridged to Brad >> Andy and Brad have conversation >> Brad then puts Andy on hold via DTMF >> Putting Andy on hold triggers a call to Carol >> When Carol answers she will be bridged with Brad whilst Andy is on hold >> Brad will then take Andy off hold via DTMF and all 3 will be connected >> Brad then hangsup and Andy and Carol are left talking until either >> hangsup >> >> -- >> Thanks Matt >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Pozdrawiam > Tomasz > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/9c950868/attachment.html From msc at freeswitch.org Tue Jan 8 04:00:09 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jan 2013 17:00:09 -0800 Subject: [Freeswitch-users] Channel variables for npdi and rn on LRN dipped calls. In-Reply-To: <50EB33ED.6010806@gmail.com> References: <50EB33ED.6010806@gmail.com> Message-ID: On Mon, Jan 7, 2013 at 12:45 PM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Ken, > > Thank you very much for feedback. I'll then follow your regex example to > parse it. > > -Victor > > On 13-01-06 12:19 PM, Ken Rice wrote: > > You have to parse out the NDPI and RN values... > > Just use a regex like > ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6});ndpi=(*.);rn=([2-9]\d{2}[2-9]\d{6})$ > That will get dialed digits in $1, ndpi value in $2, then LRN in $3, > > Just a quick FYI... be careful when you use .* because it will try to match EVERYTHING. Also, Ken accidentally typed (*.) which I'm sure wasn't what he was aiming for. ;) Maybe this would be more effective: ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6});ndpi=(.*?);rn=([2-9]\d{2}[2-9]\d{6})$ -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/1649bd26/attachment.html From krice at freeswitch.org Tue Jan 8 04:05:02 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 07 Jan 2013 19:05:02 -0600 Subject: [Freeswitch-users] Channel variables for npdi and rn on LRN dipped calls. In-Reply-To: Message-ID: Ooops yeah I mean .*... The ? Is pretty useless there... Since there should always be something if they send ndpi= On 1/7/13 7:00 PM, "Michael Collins" wrote: >>> > Just a quick FYI... be careful when you use .* because it will try to match > EVERYTHING. Also, Ken accidentally typed (*.) which I'm sure wasn't what he > was aiming for. ;) > > Maybe this would be more effective: > ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6});ndpi=(.*?);rn=([2-9]\d{2}[2-9]\d{6})$ > > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/98974ab8/attachment.html From msc at freeswitch.org Tue Jan 8 04:59:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 Jan 2013 17:59:37 -0800 Subject: [Freeswitch-users] Channel variables for npdi and rn on LRN dipped calls. In-Reply-To: References: Message-ID: In this case the ? modifies the *, so there's .* (greedy) and .*? (not greedy) You want your .* to match as little as possible instead of as much as possible. If you use .* instead of .*? then your $2 would contain both the ndpi info but also it would have ";rn=18005551212" b/c the .* would grab all that stuff. The .*? would only grab up to the ";rn". Just adding that info for posterity's sake. :) -MC On Mon, Jan 7, 2013 at 5:05 PM, Ken Rice wrote: > Ooops yeah I mean .*... The ? Is pretty useless there... Since there > should always be something if they send ndpi= > > > On 1/7/13 7:00 PM, "Michael Collins" wrote: > > > Just a quick FYI... be careful when you use .* because it will try to > match EVERYTHING. Also, Ken accidentally typed (*.) which I'm sure wasn't > what he was aiming for. ;) > > Maybe this would be more effective: > ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6});ndpi=(.*?);rn=([2-9]\d{2}[2-9]\d{6})$ > > -MC > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130107/99fc0a73/attachment-0001.html From jaybinks at gmail.com Tue Jan 8 08:27:35 2013 From: jaybinks at gmail.com (jay binks) Date: Tue, 8 Jan 2013 15:27:35 +1000 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: References: Message-ID: So what would i have to do to win that :) ( yea I setup lenny, and kinda look after him when I can find time ) I kinda knew about it but discounted it since im not in the USA. On 6 January 2013 19:04, Gabriel Gunderson wrote: > Will someone please forward this valuable infomation to Lenny? > > http://money.cnn.com/2013/01/02/pf/ftc-robocalls/index.html > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/f92f5f22/attachment.html From matt at inveroak.com Tue Jan 8 11:13:03 2013 From: matt at inveroak.com (Matt Broad) Date: Tue, 8 Jan 2013 08:13:03 +0000 Subject: [Freeswitch-users] Fwd: freeswitch on hold In-Reply-To: References: Message-ID: thanks Michael & Tomasz, I will give this a go. Is there a way of changing the key that is pressed from *4 to something else? thanks Matt On 8 January 2013 00:48, Michael Collins wrote: > Just as an FYI, the *4 is talking about the bind meta app that is found in > the vanilla configs, Local_Extension, in conf/dialplan/default.xml. Keep in > mind that this only works when the person pressing *4 is the recipient > (i.e. the b-leg). Be sure to set up a separate extension if you want the > a-leg to be able to dial *4. (Don't forget, the "default" dialplan is an > example, so be sure to review it and make appropriate changes.) > > -MC > > > On Mon, Jan 7, 2013 at 1:40 PM, Tomasz Szuster wrote: > >> Hi Matt, >> >> Some time ago I've the same questions. >> The answer is *4. >> After you press this you can bridge two separate calls Brad and Carol. >> >> If you have 3 people on the same line then maybe you have to create >> converence, don't know this. >> >> But start with *4 :) >> >> More experienced users will help you with this. >> >> Regards. >> Tom. >> >> On Mon, Jan 7, 2013 at 6:28 PM, Matt Broad wrote: >> >>> Hi >>> >>> I am wondering the best way to go about putting the below service >>> together: >>> >>> Andy calls a number >>> Andy's call is bridged to Brad >>> Andy and Brad have conversation >>> Brad then puts Andy on hold via DTMF >>> Putting Andy on hold triggers a call to Carol >>> When Carol answers she will be bridged with Brad whilst Andy is on hold >>> Brad will then take Andy off hold via DTMF and all 3 will be connected >>> Brad then hangsup and Andy and Carol are left talking until either >>> hangsup >>> >>> -- >>> Thanks Matt >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Pozdrawiam >> Tomasz >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thanks Matt This email and any attachments to it are confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of InverOak Limited. If you are not the intended recipient of this email, you must neither take any action based upon its contents, nor copy or show it to anyone. Please contact the sender if you believe you have received this email in error. This email including any attachments cannot be guaranteed to be 100% secure or error-free as information could be intercepted, corrupted, lost, destroyed, out-dated, or containing viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. InverOak Limited is a company registered in England & Wales under company number 04529594, whose registered address is Old Barn house, 2 Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/2d47fcd0/attachment.html From gerald.weber at besharp.at Tue Jan 8 12:18:33 2013 From: gerald.weber at besharp.at (Gerald Weber) Date: Tue, 8 Jan 2013 09:18:33 +0000 Subject: [Freeswitch-users] Question on Sevens Erlang Page Message-ID: Hi, i was looking fort he page http://www.dujinfang.com/past/2010/4/22/build-a-complex-hence-powerful-freeswitch-ivr-in-erlang The page seems to be offline for maintenance for weeks now and i'm not able to find it either in googles cache or wayback.archive.org. does anyone have a copy of hat page (seven maybe?) thanks in advance, gw Ps: what happened to this page ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/073c0fcd/attachment.html From tomasz.szuster at gmail.com Tue Jan 8 12:54:35 2013 From: tomasz.szuster at gmail.com (Tomasz Szuster) Date: Tue, 8 Jan 2013 10:54:35 +0100 Subject: [Freeswitch-users] Fwd: freeswitch on hold In-Reply-To: References: Message-ID: Yes, search in the dialplan.xml On 8 Jan 2013 09:18, "Matt Broad" wrote: > thanks Michael & Tomasz, I will give this a go. Is there a way of > changing the key that is pressed from *4 to something else? > > thanks > Matt > > > On 8 January 2013 00:48, Michael Collins wrote: > >> Just as an FYI, the *4 is talking about the bind meta app that is found >> in the vanilla configs, Local_Extension, in conf/dialplan/default.xml. Keep >> in mind that this only works when the person pressing *4 is the recipient >> (i.e. the b-leg). Be sure to set up a separate extension if you want the >> a-leg to be able to dial *4. (Don't forget, the "default" dialplan is an >> example, so be sure to review it and make appropriate changes.) >> >> -MC >> >> >> On Mon, Jan 7, 2013 at 1:40 PM, Tomasz Szuster wrote: >> >>> Hi Matt, >>> >>> Some time ago I've the same questions. >>> The answer is *4. >>> After you press this you can bridge two separate calls Brad and Carol. >>> >>> If you have 3 people on the same line then maybe you have to create >>> converence, don't know this. >>> >>> But start with *4 :) >>> >>> More experienced users will help you with this. >>> >>> Regards. >>> Tom. >>> >>> On Mon, Jan 7, 2013 at 6:28 PM, Matt Broad wrote: >>> >>>> Hi >>>> >>>> I am wondering the best way to go about putting the below service >>>> together: >>>> >>>> Andy calls a number >>>> Andy's call is bridged to Brad >>>> Andy and Brad have conversation >>>> Brad then puts Andy on hold via DTMF >>>> Putting Andy on hold triggers a call to Carol >>>> When Carol answers she will be bridged with Brad whilst Andy is on hold >>>> Brad will then take Andy off hold via DTMF and all 3 will be connected >>>> Brad then hangsup and Andy and Carol are left talking until either >>>> hangsup >>>> >>>> -- >>>> Thanks Matt >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Pozdrawiam >>> Tomasz >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Thanks > Matt > > This email and any attachments to it are confidential and are intended > solely for the use of the individual to whom it is addressed. Any views or > opinions expressed are solely those of the author and do not necessarily > represent those of InverOak Limited. > > If you are not the intended recipient of this email, you must neither take > any action based upon its contents, nor copy or show it to anyone. Please > contact the sender if you believe you have received this email in error. > > This email including any attachments cannot be guaranteed to be 100% > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, out-dated, or containing viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message which arise as a result of email transmission. > > InverOak Limited is a company registered in England & Wales under company > number 04529594, whose registered address is Old Barn house, 2 Wannions > Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/a113fc8d/attachment-0001.html From matt at inveroak.com Tue Jan 8 13:00:52 2013 From: matt at inveroak.com (Matt Broad) Date: Tue, 8 Jan 2013 10:00:52 +0000 Subject: [Freeswitch-users] Fwd: freeswitch on hold In-Reply-To: References: Message-ID: ok thanks for the quick response :) thanks Matt On 8 January 2013 09:54, Tomasz Szuster wrote: > Yes, search in the dialplan.xml > On 8 Jan 2013 09:18, "Matt Broad" wrote: > >> thanks Michael & Tomasz, I will give this a go. Is there a way of >> changing the key that is pressed from *4 to something else? >> >> thanks >> Matt >> >> >> On 8 January 2013 00:48, Michael Collins wrote: >> >>> Just as an FYI, the *4 is talking about the bind meta app that is found >>> in the vanilla configs, Local_Extension, in conf/dialplan/default.xml. Keep >>> in mind that this only works when the person pressing *4 is the recipient >>> (i.e. the b-leg). Be sure to set up a separate extension if you want the >>> a-leg to be able to dial *4. (Don't forget, the "default" dialplan is an >>> example, so be sure to review it and make appropriate changes.) >>> >>> -MC >>> >>> >>> On Mon, Jan 7, 2013 at 1:40 PM, Tomasz Szuster >> > wrote: >>> >>>> Hi Matt, >>>> >>>> Some time ago I've the same questions. >>>> The answer is *4. >>>> After you press this you can bridge two separate calls Brad and Carol. >>>> >>>> If you have 3 people on the same line then maybe you have to create >>>> converence, don't know this. >>>> >>>> But start with *4 :) >>>> >>>> More experienced users will help you with this. >>>> >>>> Regards. >>>> Tom. >>>> >>>> On Mon, Jan 7, 2013 at 6:28 PM, Matt Broad wrote: >>>> >>>>> Hi >>>>> >>>>> I am wondering the best way to go about putting the below service >>>>> together: >>>>> >>>>> Andy calls a number >>>>> Andy's call is bridged to Brad >>>>> Andy and Brad have conversation >>>>> Brad then puts Andy on hold via DTMF >>>>> Putting Andy on hold triggers a call to Carol >>>>> When Carol answers she will be bridged with Brad whilst Andy is on hold >>>>> Brad will then take Andy off hold via DTMF and all 3 will be connected >>>>> Brad then hangsup and Andy and Carol are left talking until either >>>>> hangsup >>>>> >>>>> -- >>>>> Thanks Matt >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Pozdrawiam >>>> Tomasz >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Thanks >> Matt >> >> This email and any attachments to it are confidential and are intended >> solely for the use of the individual to whom it is addressed. Any views or >> opinions expressed are solely those of the author and do not necessarily >> represent those of InverOak Limited. >> >> If you are not the intended recipient of this email, you must neither >> take any action based upon its contents, nor copy or show it to anyone. >> Please contact the sender if you believe you have received this email in >> error. >> >> This email including any attachments cannot be guaranteed to be 100% >> secure or error-free as information could be intercepted, corrupted, lost, >> destroyed, out-dated, or containing viruses. The sender therefore does not >> accept liability for any errors or omissions in the contents of this >> message which arise as a result of email transmission. >> >> InverOak Limited is a company registered in England & Wales under company >> number 04529594, whose registered address is Old Barn house, 2 Wannions >> Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thanks Matt This email and any attachments to it are confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of InverOak Limited. If you are not the intended recipient of this email, you must neither take any action based upon its contents, nor copy or show it to anyone. Please contact the sender if you believe you have received this email in error. This email including any attachments cannot be guaranteed to be 100% secure or error-free as information could be intercepted, corrupted, lost, destroyed, out-dated, or containing viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. InverOak Limited is a company registered in England & Wales under company number 04529594, whose registered address is Old Barn house, 2 Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/7ba02dec/attachment.html From dgarcia at anew.com.ve Tue Jan 8 16:25:27 2013 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 08 Jan 2013 08:55:27 -0430 Subject: [Freeswitch-users] Question on Sevens Erlang Page In-Reply-To: References: Message-ID: <50EC1E47.5050409@anew.com.ve> Hi, Check this link: http://translate.google.co.ve/translate?hl=es-419&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search%3Fq%3Dbuild%2Ba%2Bcomplex%2Bhence%2Bpowerful%2Bfreeswitch%2Bivr%2Bin%2Berlang%26hl%3Des-419%26client%3Dfirefox-a%26hs%3D06u%26tbo%3Dd%26rls%3Dorg.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg Original link http://www.google.co.ve/url?sa=t&rct=j&q=&esrc=s&source=web&cd=4&ved=0CEEQFjAD&url=http%3A%2F%2Fwww.freeswitch.org.cn%2F2010%2F04%2F26%2Fshi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&ei=CxzsUKu5Oo-K9QTh_ICAAg&usg=AFQjCNFxsI4KHYigeKt2ZVQsPkSrwwiW5g&bvm=bv.1357316858,d.eWU&cad=rja I dont know if the link has the same info. On 1/8/2013 4:48 AM, Gerald Weber wrote: > > Hi, > > i was looking fort he page > > http://www.dujinfang.com/past/2010/4/22/build-a-complex-hence-powerful-freeswitch-ivr-in-erlang > > > The page seems to be offline for maintenance for weeks now and i'm not > able to find it either in googles cache or wayback.archive.org. > > does anyone have a copy of hat page (seven maybe?) > > thanks in advance, > > gw > > Ps: what happened to this page ? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2221 / Virus Database: 2637/5518 - Release Date: 01/08/13 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/eb9c3e25/attachment-0001.html From mike at jerris.com Tue Jan 8 16:30:22 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Jan 2013 08:30:22 -0500 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls In-Reply-To: References: Message-ID: Speaking of lenny, those US DID's are dead. Does anyone have a US DID they can point? On Jan 8, 2013, at 12:27 AM, jay binks wrote: > So what would i have to do to win that :) > ( yea I setup lenny, and kinda look after him when I can find time ) > > I kinda knew about it but discounted it since im not in the USA. > > On 6 January 2013 19:04, Gabriel Gunderson wrote: > Will someone please forward this valuable infomation to Lenny? > > http://money.cnn.com/2013/01/02/pf/ftc-robocalls/index.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/d297a41d/attachment.html From mike at jerris.com Tue Jan 8 16:35:53 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Jan 2013 08:35:53 -0500 Subject: [Freeswitch-users] Question on Sevens Erlang Page In-Reply-To: <50EC1E47.5050409@anew.com.ve> References: <50EC1E47.5050409@anew.com.ve> Message-ID: <2B6D1002-DD1A-4F20-8128-AA80C940D29B@jerris.com> or if you want it in english: http://translate.google.co.ve/translate?hl=en&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search%3Fq%3Dbuild%2Ba%2Bcomplex%2Bhence%2Bpowerful%2Bfreeswitch%2Bivr%2Bin%2Berlang%26hl%3Des-419%26client%3Dfirefox-a%26hs%3D06u%26tbo%3Dd%26rls%3Dorg.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg On Jan 8, 2013, at 8:25 AM, Saugort Dario Garcia Tovar wrote: > Hi, > > Check this link: > http://translate.google.co.ve/translate?hl=es-419&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search%3Fq%3Dbuild%2Ba%2Bcomplex%2Bhence%2Bpowerful%2Bfreeswitch%2Bivr%2Bin%2Berlang%26hl%3Des-419%26client%3Dfirefox-a%26hs%3D06u%26tbo%3Dd%26rls%3Dorg.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg > > Original link > http://www.google.co.ve/url?sa=t&rct=j&q=&esrc=s&source=web&cd=4&ved=0CEEQFjAD&url=http%3A%2F%2Fwww.freeswitch.org.cn%2F2010%2F04%2F26%2Fshi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&ei=CxzsUKu5Oo-K9QTh_ICAAg&usg=AFQjCNFxsI4KHYigeKt2ZVQsPkSrwwiW5g&bvm=bv.1357316858,d.eWU&cad=rja > > I dont know if the link has the same info. > > > On 1/8/2013 4:48 AM, Gerald Weber wrote: >> Hi, >> >> i was looking fort he page >> http://www.dujinfang.com/past/2010/4/22/build-a-complex-hence-powerful-freeswitch-ivr-in-erlang >> >> The page seems to be offline for maintenance for weeks now and i?m not able to find it either in googles cache or wayback.archive.org. >> >> does anyone have a copy of hat page (seven maybe?) >> >> thanks in advance, >> gw >> >> Ps: what happened to this page ? >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/f4796662/attachment.html From dujinfang at gmail.com Tue Jan 8 16:50:46 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 8 Jan 2013 21:50:46 +0800 Subject: [Freeswitch-users] Question on Sevens Erlang Page In-Reply-To: <2B6D1002-DD1A-4F20-8128-AA80C940D29B@jerris.com> References: <50EC1E47.5050409@anew.com.ve> <2B6D1002-DD1A-4F20-8128-AA80C940D29B@jerris.com> Message-ID: The site was hosted on heroku but when heroku upgraded it's ruby stack it made me hard to upgrade my blog software and it was offline for a couple weeks because I was too busy to update it. I move all to github pages now so here's the new link: http://www.dujinfang.com/2010/04/22/build-a-complex-hence-powerful-freeswitch-ivr-in-erlang.html Thanks all. -- Seven Du Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, January 8, 2013 at 9:35 PM, Michael Jerris wrote: > or if you want it in english: > > http://translate.google.co.ve/translate?hl=en&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search%3Fq%3Dbuild%2Ba%2Bcomplex%2Bhence%2Bpowerful%2Bfreeswitch%2Bivr%2Bin%2Berlang%26hl%3Des-419%26client%3Dfirefox-a%26hs%3D06u%26tbo%3Dd%26rls%3Dorg.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg (http://translate.google.co.ve/translate?hl=en&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search?q=build+a+complex+hence+powerful+freeswitch+ivr+in+erlang&hl=es-419&client=firefox-a&hs=06u&tbo=d&rls=org.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg) > > > On Jan 8, 2013, at 8:25 AM, Saugort Dario Garcia Tovar wrote: > > Hi, > > > > Check this link: > > http://translate.google.co.ve/translate?hl=es-419&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search%3Fq%3Dbuild%2Ba%2Bcomplex%2Bhence%2Bpowerful%2Bfreeswitch%2Bivr%2Bin%2Berlang%26hl%3Des-419%26client%3Dfirefox-a%26hs%3D06u%26tbo%3Dd%26rls%3Dorg.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg > > > > Original link > > http://www.google.co.ve/url?sa=t&rct=j&q=&esrc=s&source=web&cd=4&ved=0CEEQFjAD&url=http%3A%2F%2Fwww.freeswitch.org.cn%2F2010%2F04%2F26%2Fshi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&ei=CxzsUKu5Oo-K9QTh_ICAAg&usg=AFQjCNFxsI4KHYigeKt2ZVQsPkSrwwiW5g&bvm=bv.1357316858,d.eWU&cad=rja > > > > I dont know if the link has the same info. > > > > > > On 1/8/2013 4:48 AM, Gerald Weber wrote: > > > Hi, > > > > > > i was looking fort he page > > > http://www.dujinfang.com/past/2010/4/22/build-a-complex-hence-powerful-freeswitch-ivr-in-erlang > > > > > > The page seems to be offline for maintenance for weeks now and i?m not able to find it either in googles cache or wayback.archive.org (http://wayback.archive.org). > > > > > > does anyone have a copy of hat page (seven maybe?) > > > > > > thanks in advance, > > > gw > > > > > > Ps: what happened to this page ? > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/c12b4f9d/attachment-0001.html From michel.brabants at gmail.com Tue Jan 8 18:22:41 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Tue, 8 Jan 2013 16:22:41 +0100 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: References: Message-ID: Hello, I'm testing freeswitch as an sbc (version 1.2.5.3). I think I discovered a bug, but I still have to verify it against git if I can find the time to compile it. Anyway, it all relates to session-timer according to me, but even if I disable session-timers, freeswitch still sends sip-update-messages to the sip-client. Freeswitch is dropping the calls after 30 seconds because of this. There is no problem with natting, firewalling, ... as far as I can tell as I'm tracing the packets on all ends. A packet from freeswitch to the sip-client (phone behind other B2B-server): UPDATE sip:10000 at lab2.ddm.priv SIP/2.0 Via: SIP/2.0/UDP 172.16.105.14;rport;branch= z9hG4bKS2pQKQ7SrNgce Max-Forwards: 70 From: ;tag=X5FKtKj9ZHaDc To: ;tag=849E1A5B-6750-4EDE-B7CF-72DF979FD126-40026 Call-ID: 5AC29923-74EE-46B2-830A-9CB78B9DB47F-39541 at 172.16.105.17 CSeq: 38492344 UPDATE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.5.3+git~20121206T050429Z~91eef34d5c Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 249 P-Asserted-Identity: "Outbound Call" <0001> v=0 o=FreeSWITCH 1357636566 1357636567 IN IP4 172.16.105.14 s=FreeSWITCH c=IN IP4 172.16.105.14 t=0 0 m=audio 20250 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 The other end of the sbc/client doesn't support session-timers or update (yate client), and it doesn't have this problem. So, the main question is: --> Why is freeswitch sending these update-messages while the codec, ... remains the same as far as I can tell. It isn't a session-timer, because it is enabled in the profile and it doesn't appear in the message too (which means the disable had some influence ...)? So, is there a way to: 1) fix the above? 2) force session-timers to use reinvite instead of update. As this case may be a way around the bug (freeswitch doing as it is the refresher while they agreed that the client is the session-timer-refresher). 3) (asked before, but not answered) Is there a way to disable sip-"update"-message-support in freeswitch? Thank you and kind regards, Michel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/76db07c5/attachment.html From sertys at gmail.com Tue Jan 8 19:29:17 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Tue, 8 Jan 2013 17:29:17 +0100 Subject: [Freeswitch-users] Freeswitch mod_sms and SIMPLE In-Reply-To: <50EB4275.2080304@quentustech.com> References: <50EB4275.2080304@quentustech.com> Message-ID: I now see i could have used more detail in my question. I am talking about mod_chat MESSAGEs which are part of the SIMPLE protocol for messaging and not mod_sms. I've stumbled upon many issues with messaging, but have little time to fiddle with it. On Jan 7, 2013 11:54 PM, "William King" wrote: > What specifically do you mean by peer to peer? There are two methods for > sending SMS messages: one that validates that the message was sent, and > one that gives a single attempt. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 01/04/2013 03:25 PM, Daniel Ivanov wrote: > > I stumbled upon this feature lately and since my clients support it, > > decided to add it. Wrote a lua wrapper over a web sms gateway and it's > > working correctly. I however found peer-to-peer chat very unreliable. > > There is no spooling of messages whenever one leg is unavailable and it > > isn't always delivering messages( i think this might be a client issue > > though). Who is maintaining the mod and what are the planned feature > > updates to it. I can write an external spooler, but i can bet there's a > > better event based approach in FS core. > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/fa2ecea6/attachment.html From matt at inveroak.com Tue Jan 8 19:56:08 2013 From: matt at inveroak.com (Matt Broad) Date: Tue, 8 Jan 2013 16:56:08 +0000 Subject: [Freeswitch-users] Fwd: freeswitch on hold In-Reply-To: References: Message-ID: Hi Tomasz, thanks for your help with this, I have managed to set this up so that I can make a call in, instigate the dial out and connect the a leg to the c leg (either hanging up or pressing * on the b leg). This works great, the only thing is the 0 press doesn't create a three way conversation as per the wiki. Pressing it has the same effect as the # key, in that it hangs up the c leg. Any ideas why that might be? thanks Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/28f87eb7/attachment.html From abaci64 at gmail.com Tue Jan 8 20:16:53 2013 From: abaci64 at gmail.com (Abaci) Date: Tue, 08 Jan 2013 12:16:53 -0500 Subject: [Freeswitch-users] Trust "Proxy-Authorization:" header Message-ID: <50EC5485.9070708@gmail.com> I would like to know if there is an option in FreeSWITCH so that if an INVITE comes in and already has the "Proxy-Authorization:" header (set on proxy), we should accept this as authenticated (and set user etc.) and not challenge it, to avoid double authentication as we already auth on the proxy. If such feature doesn't exist I would like to know if it makes sense to open a Bounty for it. Thanks From kris at kriskinc.com Tue Jan 8 22:23:11 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 8 Jan 2013 14:23:11 -0500 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Variable_ignore_display_updates On Tue, Jan 8, 2013 at 10:22 AM, Michel Brabants wrote: > Hello, > > I'm testing freeswitch as an sbc (version 1.2.5.3). I think I discovered a > bug, but I still have to verify it against git if I can find the time to > compile it. Anyway, it all relates to session-timer according to me, but > even if I disable session-timers, freeswitch still sends sip-update-messages > to the sip-client. Freeswitch is dropping the calls after 30 seconds because > of this. There is no problem with natting, firewalling, ... as far as I can > tell as I'm tracing the packets on all ends. A packet from freeswitch to the > sip-client (phone behind other B2B-server): > > UPDATE sip:10000 at lab2.ddm.priv SIP/2.0 > Via: SIP/2.0/UDP 172.16.105.14;rport;branch= > z9hG4bKS2pQKQ7SrNgce > Max-Forwards: 70 > From: ;tag=X5FKtKj9ZHaDc > To: > ;tag=849E1A5B-6750-4EDE-B7CF-72DF979FD126-40026 > Call-ID: 5AC29923-74EE-46B2-830A-9CB78B9DB47F-39541 at 172.16.105.17 > CSeq: 38492344 UPDATE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.2.5.3+git~20121206T050429Z~91eef34d5c > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, > NOTIFY > Supported: precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 249 > P-Asserted-Identity: "Outbound Call" <0001> > > v=0 > o=FreeSWITCH 1357636566 1357636567 IN IP4 172.16.105.14 > s=FreeSWITCH > c=IN IP4 172.16.105.14 > t=0 0 > m=audio 20250 RTP/AVP 9 127 > a=rtpmap:9 G722/8000 > a=rtpmap:127 telephone-event/8000 > a=fmtp:127 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > > > The other end of the sbc/client doesn't support session-timers or update > (yate client), and it doesn't have this problem. > > So, the main question is: > > --> Why is freeswitch sending these update-messages while the codec, ... > remains the same as far as I can tell. It isn't a session-timer, because it > is enabled in the profile and it doesn't appear in the message too (which > means the disable had some influence ...)? > > So, is there a way to: > 1) fix the above? > 2) force session-timers to use reinvite instead of update. As this case may > be a way around the bug (freeswitch doing as it is the refresher while they > agreed that the client is the session-timer-refresher). > 3) (asked before, but not answered) Is there a way to disable > sip-"update"-message-support in freeswitch? > > Thank you and kind regards, > > Michel > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From steveayre at gmail.com Tue Jan 8 22:40:46 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 8 Jan 2013 19:40:46 +0000 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: References: Message-ID: Does the client send an ACK for the UPDATE? On 8 January 2013 15:22, Michel Brabants wrote: > Hello, > > I'm testing freeswitch as an sbc (version 1.2.5.3). I think I discovered a > bug, but I still have to verify it against git if I can find the time to > compile it. Anyway, it all relates to session-timer according to me, but > even if I disable session-timers, freeswitch still sends sip-update-messages > to the sip-client. Freeswitch is dropping the calls after 30 seconds because > of this. There is no problem with natting, firewalling, ... as far as I can > tell as I'm tracing the packets on all ends. A packet from freeswitch to the > sip-client (phone behind other B2B-server): > > UPDATE sip:10000 at lab2.ddm.priv SIP/2.0 > Via: SIP/2.0/UDP 172.16.105.14;rport;branch= > z9hG4bKS2pQKQ7SrNgce > Max-Forwards: 70 > From: ;tag=X5FKtKj9ZHaDc > To: > ;tag=849E1A5B-6750-4EDE-B7CF-72DF979FD126-40026 > Call-ID: 5AC29923-74EE-46B2-830A-9CB78B9DB47F-39541 at 172.16.105.17 > CSeq: 38492344 UPDATE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.2.5.3+git~20121206T050429Z~91eef34d5c > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, > NOTIFY > Supported: precondition, path, replaces > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 249 > P-Asserted-Identity: "Outbound Call" <0001> > > v=0 > o=FreeSWITCH 1357636566 1357636567 IN IP4 172.16.105.14 > s=FreeSWITCH > c=IN IP4 172.16.105.14 > t=0 0 > m=audio 20250 RTP/AVP 9 127 > a=rtpmap:9 G722/8000 > a=rtpmap:127 telephone-event/8000 > a=fmtp:127 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > > > The other end of the sbc/client doesn't support session-timers or update > (yate client), and it doesn't have this problem. > > So, the main question is: > > --> Why is freeswitch sending these update-messages while the codec, ... > remains the same as far as I can tell. It isn't a session-timer, because it > is enabled in the profile and it doesn't appear in the message too (which > means the disable had some influence ...)? > > So, is there a way to: > 1) fix the above? > 2) force session-timers to use reinvite instead of update. As this case may > be a way around the bug (freeswitch doing as it is the refresher while they > agreed that the client is the session-timer-refresher). > 3) (asked before, but not answered) Is there a way to disable > sip-"update"-message-support in freeswitch? > > Thank you and kind regards, > > Michel > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From infos at madovsky.org Tue Jan 8 22:13:45 2013 From: infos at madovsky.org (Madovsky) Date: Tue, 8 Jan 2013 11:13:45 -0800 (PST) Subject: [Freeswitch-users] mod_voicemail compilation error In-Reply-To: References: <592A9CF93E12394E8472A6CC66E66BF2337901@Mail-Kilo.squay.com> <1FFF97C269757C458224B7C895F35F151CAFE9@cantor.std.visionutv.se> <592A9CF93E12394E8472A6CC66E66BF2337941@Mail-Kilo.squay.com> <592A9CF93E12394E8472A6CC66E66BF2338E30@Mail-Kilo.squay.com> Message-ID: <1357672425805-7586126.post@n2.nabble.com> this thread is a little bit outdated but I faced of the same compilation problem on Fedora 10 64bits. I had to compile voicemail and voicemail_ivr after compiled FS. Used January 8 git source -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-voicemail-compilation-error-tp7584862p7586126.html Sent from the freeswitch-users mailing list archive at Nabble.com. From kris at kriskinc.com Tue Jan 8 23:46:22 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 8 Jan 2013 15:46:22 -0500 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: References: Message-ID: The correct positive response to an UPDATE would be a 200 that is then ACKd. See RFC 3311: http://tools.ietf.org/html/rfc3311 On Tue, Jan 8, 2013 at 2:40 PM, Steven Ayre wrote: > Does the client send an ACK for the UPDATE? > -- Kristian Kielhofner From schoch+freeswitch.org at xwin32.com Wed Jan 9 00:05:30 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 8 Jan 2013 13:05:30 -0800 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: <50DE75C4.1030500@communicatefreely.net> References: <50DE75C4.1030500@communicatefreely.net> Message-ID: On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < fs-list at communicatefreely.net> wrote: > Hi Steven, > > I would recommend using a proper domain name as much as possible. For > one, it looks > nicer! A SIP URI is supposed to be user at domain like an e-mail address > is, and I hope that > one day URI dialing will be common place, so we might as well do it right > the first time. > What you're saying is that "domain" should really be a fully-qualified host name that points via DNS to the actual host on which FreeSwitch is running. That is, the domain should be "pbx.example.com" instead of just " example.com", as the last example would most likely point to a web server, not the SIP server. Do I have that right? Next, in the configuration for Polycom phones (for example), there are 2 fields that both have the userid. In the example in http://wiki.freeswitch.org/wiki/Polycom_configuration it has: reg.1.auth.userId="1000" and reg.1.address="1000 at fs.domain.local" How is the "address" value used? Is that sent in the SIP registration message? If that's the case, what does Freeswitch do with it? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/1c4013dc/attachment.html From steveayre at gmail.com Wed Jan 9 00:11:19 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 8 Jan 2013 21:11:19 +0000 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: References: Message-ID: Yep my mistake... rather I meant is FS getting a response? (Wondering if FS is retransmitting and/or timing out the call on no response since you say it hangs up 30s later) On 8 January 2013 20:46, Kristian Kielhofner wrote: > The correct positive response to an UPDATE would be a 200 that is then > ACKd. See RFC 3311: > > http://tools.ietf.org/html/rfc3311 > > On Tue, Jan 8, 2013 at 2:40 PM, Steven Ayre wrote: >> Does the client send an ACK for the UPDATE? >> > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lconroy at insensate.co.uk Wed Jan 9 00:56:02 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Tue, 8 Jan 2013 21:56:02 +0000 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: References: <50DE75C4.1030500@communicatefreely.net> Message-ID: Hi there, at the risk of butting in on someone else's party ... Nope; your interpretations is NOT best practice. I have some sympathy, as the term domain is overloaded within fS. A sip address consists of a userpart and a domain part -- e.g., The sip domain is similar to an email domain -- e.g., With email, you need to do a lookup of the MX record in DNS to find the FQDN of the machine that handles mail for the domain. With SIP (see RFC 3263), you do a lookup on the SRV record (at _sip._udp.) to find the machine that handles SIP registrations/incalls for the domain. That also gives you the port on which that machine is listening. (Yup, you can also have a NAPTR record in the domain to tell you where the SRV record is, but many folks don't bother -- for Best Practice, you should, but ...) There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) support that means most SIP clients will look for the SRV and, if it can't be found (or there's an IP address rather than a DNS -style domain, in which case the SIP client won't bother hunting the SRV), the client will guess that the domain has a machine (i.e. it will look for an A or AAAA record), and also guess it's listening on 5060 (the default port). Email is the same (mail to fred at example.com, and strictly the sender will do a check for a MX and then look for an A record for example.com, and try there). However, relying on that default "get out" clause is definitely NOT what you should do for BCP. Using the hostname as the sip domain is a kludge -- the FQDN with A record usually works, but it's not what you want to do. SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at _sip._udp., and you're done. No need for an A record at that domain at all. (RFC 3263 is not too hard to read, for a change -- it's certainly shorter than RFC 3261, and it even has an ASCII art diagram :). all the best, Lawrence On 8 Jan 2013, at 21:05, Steven Schoch wrote: > On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < > fs-list at communicatefreely.net> wrote: > >> Hi Steven, >> >> I would recommend using a proper domain name as much as possible. For >> one, it looks >> nicer! A SIP URI is supposed to be user at domain like an e-mail address >> is, and I hope that >> one day URI dialing will be common place, so we might as well do it right >> the first time. >> > > What you're saying is that "domain" should really be a fully-qualified host > name that points via DNS to the actual host on which FreeSwitch is running. > That is, the domain should be "pbx.example.com" instead of just " > example.com", as the last example would most likely point to a web server, > not the SIP server. Do I have that right? > > Next, in the configuration for Polycom phones (for example), there are 2 > fields that both have the userid. In the example in > http://wiki.freeswitch.org/wiki/Polycom_configuration it has: > > reg.1.auth.userId="1000" > > and > > reg.1.address="1000 at fs.domain.local" > > How is the "address" value used? Is that sent in the SIP registration > message? If that's the case, what does Freeswitch do with it? > > -- > Steve > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Jan 9 01:56:42 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Jan 2013 14:56:42 -0800 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: References: <50DE75C4.1030500@communicatefreely.net> Message-ID: Lawrence, Thanks for this explanation. It was very well written. I'm looking for a place to codify this on the wiki so that it gets preserved... :) -MC On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy wrote: > Hi there, > at the risk of butting in on someone else's party ... > Nope; your interpretations is NOT best practice. > I have some sympathy, as the term domain is overloaded within fS. > > A sip address consists of a userpart and a domain part -- e.g., > > The sip domain is similar to an email domain -- e.g., user at maildomain> > With email, you need to do a lookup of the MX record in DNS to find the > FQDN of the machine that handles mail for the domain. > With SIP (see RFC 3263), you do a lookup on the SRV record (at > _sip._udp.) to find the machine that handles SIP > registrations/incalls for the domain. That also gives you the port on > which that machine is listening. > (Yup, you can also have a NAPTR record in the domain to tell you where the > SRV record is, but many folks don't bother -- for Best Practice, you > should, but ...) > > There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) > support that means most SIP clients will look for the SRV and, if it can't > be found (or there's an IP address rather than a DNS -style domain, in > which case the SIP client won't bother hunting the SRV), the client will > guess that the domain has a machine (i.e. it will look for an A or AAAA > record), and also guess it's listening on 5060 (the default port). > Email is the same (mail to fred at example.com, and strictly the sender will > do a check for a MX and then look for an A record for example.com, and > try there). > > However, relying on that default "get out" clause is definitely NOT what > you should do for BCP. > Using the hostname as the sip domain is a kludge -- the FQDN with A record > usually works, but it's not what you want to do. > > SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at > _sip._udp., and you're done. No need for an A record at that domain > at all. > > (RFC 3263 is not too hard to read, for a change -- it's certainly shorter > than RFC 3261, and it even has an ASCII art diagram :). > > all the best, > Lawrence > > On 8 Jan 2013, at 21:05, Steven Schoch wrote: > > > On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < > > fs-list at communicatefreely.net> wrote: > > > >> Hi Steven, > >> > >> I would recommend using a proper domain name as much as possible. For > >> one, it looks > >> nicer! A SIP URI is supposed to be user at domain like an e-mail address > >> is, and I hope that > >> one day URI dialing will be common place, so we might as well do it > right > >> the first time. > >> > > > > What you're saying is that "domain" should really be a fully-qualified > host > > name that points via DNS to the actual host on which FreeSwitch is > running. > > That is, the domain should be "pbx.example.com" instead of just " > > example.com", as the last example would most likely point to a web > server, > > not the SIP server. Do I have that right? > > > > Next, in the configuration for Polycom phones (for example), there are 2 > > fields that both have the userid. In the example in > > http://wiki.freeswitch.org/wiki/Polycom_configuration it has: > > > > reg.1.auth.userId="1000" > > > > and > > > > reg.1.address="1000 at fs.domain.local" > > > > How is the "address" value used? Is that sent in the SIP registration > > message? If that's the case, what does Freeswitch do with it? > > > > -- > > Steve > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/7e02d07a/attachment-0001.html From schoch+freeswitch.org at xwin32.com Wed Jan 9 02:02:53 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 8 Jan 2013 15:02:53 -0800 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: References: <50DE75C4.1030500@communicatefreely.net> Message-ID: On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy wrote: > Hi there, > at the risk of butting in on someone else's party ... > Nope; your interpretations is NOT best practice. > This party is open to all. I appreciate these answers, and as usual I have noticed that the fastest way to learn is to do it wrong, and then have somebody correct me! Thanks! > With SIP (see RFC 3263), you do a lookup on the SRV record (at > _sip._udp.) to find the machine that handles SIP > registrations/incalls for the domain. That also gives you the port on > which that machine is listening. > For example, I did a lookup: $ dig SRV _sip._udp.sip.flowroute.com. This returns: _sip._udp.sip.flowroute.com. 43200 IN SRV 20 10 5060 sip-ca1.flowroute.com. _sip._udp.sip.flowroute.com. 43200 IN SRV 10 10 5060 sip-nv1.flowroute.com. > (Yup, you can also have a NAPTR record in the domain to tell you where the > SRV record is, but many folks don't bother -- for Best Practice, you > should, but ...) > $ dig NAPTR sip.flowroute.com sip.flowroute.com. 42818 IN NAPTR 100 10 "s" "SIP+D2U" "" _sip._udp.sip.flowroute.com. sip.flowroute.com. 42818 IN NAPTR 102 20 "s" "SIP+D2U" "" _sip._tcp.sip.flowroute.com. I found that _sip._tcp.sip.flowroute.com does not have a SRV record. I don't know what that means. The RFC says that you should use "SIP+D2T" for TCP. Now, as this relates to the Polycom phone example, there are these fields: reg.1.address="1000 at fs.domain.local" reg.1.auth.userId="1000" reg.1.server.1.address="10.10.10.11" reg.1.server.1.port="5060" Since this is a VoIP phone that loads its configuration from the server, not VoIP software that is configured by the user, it really only needs the IP address, so the server address field is kind of moot. But the example domain here is "fs.domain.local". Why not just "domain.local"? Why didn't Flowroute use "flowroute.com" instead of " sip.flowroute.com"? And my other question was how is the address "1000 at fs.domain.local" used? Does FreeSwitch use that or does it just use the auth.userId? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/01380d58/attachment.html From msc at freeswitch.org Wed Jan 9 02:21:23 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Jan 2013 15:21:23 -0800 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: References: <50DE75C4.1030500@communicatefreely.net> Message-ID: I couldn't stand it, so I added Lawrence's info to this page: http://wiki.freeswitch.org/wiki/DNS#Explanation_of_DNS.2FSRV.2FNAPTR Cheers, MC On Tue, Jan 8, 2013 at 2:56 PM, Michael Collins wrote: > Lawrence, > > Thanks for this explanation. It was very well written. I'm looking for a > place to codify this on the wiki so that it gets preserved... :) > > -MC > > > On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy wrote: > >> Hi there, >> at the risk of butting in on someone else's party ... >> Nope; your interpretations is NOT best practice. >> I have some sympathy, as the term domain is overloaded within fS. >> >> A sip address consists of a userpart and a domain part -- e.g., >> >> The sip domain is similar to an email domain -- e.g., > user at maildomain> >> With email, you need to do a lookup of the MX record in DNS to find the >> FQDN of the machine that handles mail for the domain. >> With SIP (see RFC 3263), you do a lookup on the SRV record (at >> _sip._udp.) to find the machine that handles SIP >> registrations/incalls for the domain. That also gives you the port on >> which that machine is listening. >> (Yup, you can also have a NAPTR record in the domain to tell you where >> the SRV record is, but many folks don't bother -- for Best Practice, you >> should, but ...) >> >> There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) >> support that means most SIP clients will look for the SRV and, if it can't >> be found (or there's an IP address rather than a DNS -style domain, in >> which case the SIP client won't bother hunting the SRV), the client will >> guess that the domain has a machine (i.e. it will look for an A or AAAA >> record), and also guess it's listening on 5060 (the default port). >> Email is the same (mail to fred at example.com, and strictly the sender >> will do a check for a MX and then look for an A record for example.com, >> and try there). >> >> However, relying on that default "get out" clause is definitely NOT what >> you should do for BCP. >> Using the hostname as the sip domain is a kludge -- the FQDN with A >> record usually works, but it's not what you want to do. >> >> SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV >> at _sip._udp., and you're done. No need for an A record at that >> domain at all. >> >> (RFC 3263 is not too hard to read, for a change -- it's certainly shorter >> than RFC 3261, and it even has an ASCII art diagram :). >> >> all the best, >> Lawrence >> >> On 8 Jan 2013, at 21:05, Steven Schoch wrote: >> >> > On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < >> > fs-list at communicatefreely.net> wrote: >> > >> >> Hi Steven, >> >> >> >> I would recommend using a proper domain name as much as possible. For >> >> one, it looks >> >> nicer! A SIP URI is supposed to be user at domain like an e-mail address >> >> is, and I hope that >> >> one day URI dialing will be common place, so we might as well do it >> right >> >> the first time. >> >> >> > >> > What you're saying is that "domain" should really be a fully-qualified >> host >> > name that points via DNS to the actual host on which FreeSwitch is >> running. >> > That is, the domain should be "pbx.example.com" instead of just " >> > example.com", as the last example would most likely point to a web >> server, >> > not the SIP server. Do I have that right? >> > >> > Next, in the configuration for Polycom phones (for example), there are 2 >> > fields that both have the userid. In the example in >> > http://wiki.freeswitch.org/wiki/Polycom_configuration it has: >> > >> > reg.1.auth.userId="1000" >> > >> > and >> > >> > reg.1.address="1000 at fs.domain.local" >> > >> > How is the "address" value used? Is that sent in the SIP registration >> > message? If that's the case, what does Freeswitch do with it? >> > >> > -- >> > Steve >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/cd182ebc/attachment.html From lconroy at insensate.co.uk Wed Jan 9 02:35:58 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Tue, 8 Jan 2013 23:35:58 +0000 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: References: <50DE75C4.1030500@communicatefreely.net> Message-ID: <7D1B08FC-5E9C-4E58-A9BC-B68968EB0187@insensate.co.uk> Hi Michael, folks, if you're going to codify it, I did slightly simplify things: (full disclosure -- I disagreed with it at the time which is probably why I forgot to mention it -- honest). There IS a wrinkle to this special case: If you have a URL of the form the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host example.com (i.e., there's a machine called example.com, and it is handling SIP traffic for that domainpart). Basically, if you see a colon in the domainpart, you're looking for a machine -- otherwise you're looking for a NAPTR (and/or a SRV at _sip._udp.). I'd put that before the paragraph starting "However, relying on ..." Curiously enough, the old 2543-compliant servers did hunt the SRV rather than giving up and looking for an A, so this was a change. Such fun was had re-writing implementations (plural) and testing them yet again. Sigh. all the best, Lawrence On 8 Jan 2013, at 22:56, Michael Collins wrote: > Lawrence, > > Thanks for this explanation. It was very well written. I'm looking for a > place to codify this on the wiki so that it gets preserved... :) > > -MC > > On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy wrote: > >> Hi there, >> at the risk of butting in on someone else's party ... >> Nope; your interpretations is NOT best practice. >> I have some sympathy, as the term domain is overloaded within fS. >> >> A sip address consists of a userpart and a domain part -- e.g., >> >> The sip domain is similar to an email domain -- e.g., > user at maildomain> >> With email, you need to do a lookup of the MX record in DNS to find the >> FQDN of the machine that handles mail for the domain. >> With SIP (see RFC 3263), you do a lookup on the SRV record (at >> _sip._udp.) to find the machine that handles SIP >> registrations/incalls for the domain. That also gives you the port on >> which that machine is listening. >> (Yup, you can also have a NAPTR record in the domain to tell you where the >> SRV record is, but many folks don't bother -- for Best Practice, you >> should, but ...) >> >> There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) >> support that means most SIP clients will look for the SRV and, if it can't >> be found (or there's an IP address rather than a DNS -style domain, in >> which case the SIP client won't bother hunting the SRV), the client will >> guess that the domain has a machine (i.e. it will look for an A or AAAA >> record), and also guess it's listening on 5060 (the default port). >> Email is the same (mail to fred at example.com, and strictly the sender will >> do a check for a MX and then look for an A record for example.com, and >> try there). >> >> However, relying on that default "get out" clause is definitely NOT what >> you should do for BCP. >> Using the hostname as the sip domain is a kludge -- the FQDN with A record >> usually works, but it's not what you want to do. >> >> SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at >> _sip._udp., and you're done. No need for an A record at that domain >> at all. >> >> (RFC 3263 is not too hard to read, for a change -- it's certainly shorter >> than RFC 3261, and it even has an ASCII art diagram :). >> >> all the best, >> Lawrence >> >> On 8 Jan 2013, at 21:05, Steven Schoch wrote: >> >>> On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < >>> fs-list at communicatefreely.net> wrote: >>> >>>> Hi Steven, >>>> >>>> I would recommend using a proper domain name as much as possible. For >>>> one, it looks >>>> nicer! A SIP URI is supposed to be user at domain like an e-mail address >>>> is, and I hope that >>>> one day URI dialing will be common place, so we might as well do it >> right >>>> the first time. >>>> >>> >>> What you're saying is that "domain" should really be a fully-qualified >> host >>> name that points via DNS to the actual host on which FreeSwitch is >> running. >>> That is, the domain should be "pbx.example.com" instead of just " >>> example.com", as the last example would most likely point to a web >> server, >>> not the SIP server. Do I have that right? >>> >>> Next, in the configuration for Polycom phones (for example), there are 2 >>> fields that both have the userid. In the example in >>> http://wiki.freeswitch.org/wiki/Polycom_configuration it has: >>> >>> reg.1.auth.userId="1000" >>> >>> and >>> >>> reg.1.address="1000 at fs.domain.local" >>> >>> How is the "address" value used? Is that sent in the SIP registration >>> message? If that's the case, what does Freeswitch do with it? >>> >>> -- >>> Steve >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lconroy at insensate.co.uk Wed Jan 9 03:04:41 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Wed, 9 Jan 2013 00:04:41 +0000 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: References: <50DE75C4.1030500@communicatefreely.net> Message-ID: <1ADA846F-0D19-4FE3-AFA3-7CA5E8E4C127@insensate.co.uk> Hi again Steven, with apologies for top-posting ... Re. flowroute.com Many of us have sup-domains for different classes of users -- looks like your sub-domain is sip.flowroute.com. However, flowroute.com does have a NAPTR (it has two): ;; ANSWER SECTION: flowroute.com. 21600 IN NAPTR 100 10 "s" "SIP+D2U" "" _sip._udp.flowroute.com. flowroute.com. 21600 IN NAPTR 102 20 "s" "SIP+D2T" "" _sip._tcp.flowroute.com. Also, the _sip._udp.flowroute SRV shows: ;; ANSWER SECTION: _sip._udp.flowroute.com. 43200 IN SRV 20 10 5060 sip-ca1.flowroute.com. _sip._udp.flowroute.com. 43200 IN SRV 10 10 5060 sip-nv1.flowroute.com. as does the _sip.tcp.flowroute.com SRV: ;; ANSWER SECTION: _sip._tcp.flowroute.com. 43200 IN SRV 10 10 5060 sip-nv1.flowroute.com. _sip._tcp.flowroute.com. 43200 IN SRV 20 10 5060 sip-ca1.flowroute.com. so -- if a client is trying to call to sip:1000 at flowroute.com, that client looks up the NAPTR set for flowroute.com (the sip domainpart), gets back the two entries, and chooses to look for a SRV under _sip._udp.flowroute.com (if it wants to call using UDP) or _sip.tcp.flowroute.com (if it wants to call using TCP or use sips:). flowroute would prefer that you contacted them by UDP (it has a lower preference -- 10 versus the D2T's 20). If you do contact them, they'd prefer that you used sip-nv1.flowroute.com. (as that has a better/lower preference than sip-cal). On to the polycoms -- I don't have any of these, so I'm guessing, but ... It looks strongly like the registration server is at fs.domain.local, which has an IP address of 10.10.10.11 for the DNS-challenged, and is listening on port 5060. The AuthID/UserID for this 'phone to use 1000, which is kinda apparent from the SIP address the phone has, which is sip:1000 at fs.domain.local. I'd AssUMe that there's also a password field :). So ... if you had mydomain.com, and for your internal use you had a sub-domain of, say, internal.mydomain.com, and you had one fS handling your local 'phones, running on a machine called fs.internal.domain.com which had an IP address of 10.10.10.11, I'd put into the local view of DNS: internal.mydomain.com. IN NAPTR 100 10 "s" "SIP+D2U" "" _sip._udp.internal.mydomain.com. _sip._udp.internal.mydomain.com. IN SRV 10 10 5060 fs.internal.mydomain.com fs.internal.mydomain.com. IN A 10.10.10.11 Note that I'm assuming that you'd be using a split-view DNS (i.e., your local DNS server would only give these answers to its local machines) as it's kinda impolite to give private addresses like 10.10.10.11 to the outside world -- clients are sometimes too dumb to realise they can't contact that server using this address. [...which is why we have one way audio -- stupid implementers :]p Hope that helps, Lawrence On 8 Jan 2013, at 23:02, Steven Schoch wrote: > On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy wrote: > >> Hi there, >> at the risk of butting in on someone else's party ... >> Nope; your interpretations is NOT best practice. >> > > This party is open to all. I appreciate these answers, and as usual I have > noticed that the fastest way to learn is to do it wrong, and then have > somebody correct me! Thanks! > > >> With SIP (see RFC 3263), you do a lookup on the SRV record (at >> _sip._udp.) to find the machine that handles SIP >> registrations/incalls for the domain. That also gives you the port on >> which that machine is listening. >> > > For example, I did a lookup: > > $ dig SRV _sip._udp.sip.flowroute.com. > > This returns: > > _sip._udp.sip.flowroute.com. 43200 IN SRV 20 10 5060 > sip-ca1.flowroute.com. > _sip._udp.sip.flowroute.com. 43200 IN SRV 10 10 5060 > sip-nv1.flowroute.com. > > > >> (Yup, you can also have a NAPTR record in the domain to tell you where the >> SRV record is, but many folks don't bother -- for Best Practice, you >> should, but ...) >> > > $ dig NAPTR sip.flowroute.com > > sip.flowroute.com. 42818 IN NAPTR 100 10 "s" "SIP+D2U" "" > _sip._udp.sip.flowroute.com. > sip.flowroute.com. 42818 IN NAPTR 102 20 "s" "SIP+D2U" "" > _sip._tcp.sip.flowroute.com. > > I found that _sip._tcp.sip.flowroute.com does not have a SRV record. I > don't know what that means. The RFC says that you should use "SIP+D2T" for > TCP. > > Now, as this relates to the Polycom phone example, there are these fields: > > reg.1.address="1000 at fs.domain.local" > reg.1.auth.userId="1000" > reg.1.server.1.address="10.10.10.11" > reg.1.server.1.port="5060" > > Since this is a VoIP phone that loads its configuration from the server, > not VoIP software that is configured by the user, it really only needs the > IP address, so the server address field is kind of moot. > > But the example domain here is "fs.domain.local". Why not just > "domain.local"? Why didn't Flowroute use "flowroute.com" instead of " > sip.flowroute.com"? > > And my other question was how is the address "1000 at fs.domain.local" used? > Does FreeSwitch use that or does it just use the auth.userId? > > -- > Steve > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lconroy at insensate.co.uk Wed Jan 9 03:25:18 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Wed, 9 Jan 2013 00:25:18 +0000 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: <1ADA846F-0D19-4FE3-AFA3-7CA5E8E4C127@insensate.co.uk> References: <50DE75C4.1030500@communicatefreely.net> <1ADA846F-0D19-4FE3-AFA3-7CA5E8E4C127@insensate.co.uk> Message-ID: Hi one last time Steven, folks, Oops -- you can't get the staff, 'nourdays ... For "fs.internal.domain.com", read "fs.internal.mydomain.com" :-? -- time for bed. all the best, Lawrence (and the earlier sup-domains should have been sub-domains :-? -- time for bed.) On 9 Jan 2013, at 00:04, Lawrence Conroy wrote: > called fs.internal.domain.com which had an IP address of 10.10.10.11, I'd put into the local view of DNS: From msc at freeswitch.org Wed Jan 9 03:29:10 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Jan 2013 16:29:10 -0800 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: <7D1B08FC-5E9C-4E58-A9BC-B68968EB0187@insensate.co.uk> References: <50DE75C4.1030500@communicatefreely.net> <7D1B08FC-5E9C-4E58-A9BC-B68968EB0187@insensate.co.uk> Message-ID: Done! Thanks, MC On Tue, Jan 8, 2013 at 3:35 PM, Lawrence Conroy wrote: > Hi Michael, folks, > if you're going to codify it, I did slightly simplify things: > (full disclosure -- I disagreed with it at the time which is probably why > I forgot to mention it -- honest). > > There IS a wrinkle to this special case: If you have a URL of the form < > sip:user at example.com:5055> the client should NOT look for a NAPTR or SRV, > but instead go straight to looking for an A or AAAA for the host > example.com (i.e., there's a machine called example.com, and it is > handling SIP traffic for that domainpart). > Basically, if you see a colon in the domainpart, you're looking for a > machine -- otherwise you're looking for a NAPTR (and/or a SRV at > _sip._udp.). > > I'd put that before the paragraph starting "However, relying on ..." > > Curiously enough, the old 2543-compliant servers did hunt the SRV rather > than giving up and looking for an A, so this was a change. Such fun was had > re-writing implementations (plural) and testing them yet again. Sigh. > > all the best, > Lawrence > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/c6a0a9ad/attachment.html From michel.brabants at gmail.com Wed Jan 9 03:38:29 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Wed, 9 Jan 2013 01:38:29 +0100 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: References: Message-ID: I think i already disabled this. This seems related to Sofia soa. Thanks. Op 8 jan. 2013 20:26 schreef "Kristian Kielhofner" het volgende: > http://wiki.freeswitch.org/wiki/Variable_ignore_display_updates > > On Tue, Jan 8, 2013 at 10:22 AM, Michel Brabants > wrote: > > Hello, > > > > I'm testing freeswitch as an sbc (version 1.2.5.3). I think I discovered > a > > bug, but I still have to verify it against git if I can find the time to > > compile it. Anyway, it all relates to session-timer according to me, but > > even if I disable session-timers, freeswitch still sends > sip-update-messages > > to the sip-client. Freeswitch is dropping the calls after 30 seconds > because > > of this. There is no problem with natting, firewalling, ... as far as I > can > > tell as I'm tracing the packets on all ends. A packet from freeswitch to > the > > sip-client (phone behind other B2B-server): > > > > UPDATE sip:10000 at lab2.ddm.priv SIP/2.0 > > Via: SIP/2.0/UDP 172.16.105.14;rport;branch= > > z9hG4bKS2pQKQ7SrNgce > > Max-Forwards: 70 > > From: ;tag=X5FKtKj9ZHaDc > > To: > > ;tag=849E1A5B-6750-4EDE-B7CF-72DF979FD126-40026 > > Call-ID: 5AC29923-74EE-46B2-830A-9CB78B9DB47F-39541 at 172.16.105.17 > > CSeq: 38492344 UPDATE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.2.5.3 > +git~20121206T050429Z~91eef34d5c > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REFER, > > NOTIFY > > Supported: precondition, path, replaces > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 249 > > P-Asserted-Identity: "Outbound Call" <0001> > > > > v=0 > > o=FreeSWITCH 1357636566 1357636567 IN IP4 172.16.105.14 > > s=FreeSWITCH > > c=IN IP4 172.16.105.14 > > t=0 0 > > m=audio 20250 RTP/AVP 9 127 > > a=rtpmap:9 G722/8000 > > a=rtpmap:127 telephone-event/8000 > > a=fmtp:127 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > > > > > > > The other end of the sbc/client doesn't support session-timers or update > > (yate client), and it doesn't have this problem. > > > > So, the main question is: > > > > --> Why is freeswitch sending these update-messages while the codec, ... > > remains the same as far as I can tell. It isn't a session-timer, because > it > > is enabled in the profile and it doesn't appear in the message too (which > > means the disable had some influence ...)? > > > > So, is there a way to: > > 1) fix the above? > > 2) force session-timers to use reinvite instead of update. As this case > may > > be a way around the bug (freeswitch doing as it is the refresher while > they > > agreed that the client is the session-timer-refresher). > > 3) (asked before, but not answered) Is there a way to disable > > sip-"update"-message-support in freeswitch? > > > > Thank you and kind regards, > > > > Michel > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/47e7d301/attachment-0001.html From muqtheear.s at gmail.com Wed Jan 9 04:33:08 2013 From: muqtheear.s at gmail.com (Muqtheear S) Date: Wed, 9 Jan 2013 07:03:08 +0530 Subject: [Freeswitch-users] Issue: Invalid codec mpeg4-generic! Message-ID: mod_mp4 module is looking for Hint tracks in the .mp4 file. "Hint tracks are a series of instructions in a mp4 container file that tell a server how to transmit packets." To create hint tracks for a mp4 file is fairly simple with the mp4creator program, we just executed the: mp4creator -hint= [-p ] As we know .mp4 has both video+audio data in it. We could also find audio and video encoding information of a .mp4 file as shown below. #mp4info minnale.mp4 mp4info version 1.5.0.1 minnale.mp4: Track Type Info 1 video MPEG-4 Simple @ L1, 97.833 secs, 136 kbps, 176x144 @ 12.000041 fps 2 audio MPEG-4 AAC LC, 97.568 secs, 64 kbps, 32000 Hz 3 hint Payload MP4V-ES for track 1 4 hint Payload mpeg4-generic for track 2 Metadata Tool: mp4creator 1.6 Now we are getting "Invalid codec mpeg4-generic!" error. Please guide me. Do we need to have mpeg4-generic codec in Freeswitch server? Thanks & Regards, Muqtheear.S -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/302eceb0/attachment-0001.html From itsusama at gmail.com Wed Jan 9 05:01:11 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Wed, 9 Jan 2013 07:01:11 +0500 Subject: [Freeswitch-users] FTC offers $50, 000 in contest to end robocalls Message-ID: Hi, You could try IPKall.com, it provides free DIDs and supports up to 240 calls/DID IIRC, I had some DTMF issues using it though. Phone.com also provides free inbound and you can do it for as many channels you want but its limited to 200 mins/month. Cheers. On Tue, Jan 8, 2013 at 6:51 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: FTC offers $50, 000 in contest to end robocalls > (Michael Jerris) > 2. Re: Question on Sevens Erlang Page (Michael Jerris) > 3. Re: Question on Sevens Erlang Page (Seven Du) > > > ---------- Forwarded message ---------- > From: Michael Jerris > To: FreeSWITCH Users Help > Cc: > Date: Tue, 8 Jan 2013 08:30:22 -0500 > Subject: Re: [Freeswitch-users] FTC offers $50, 000 in contest to end > robocalls > Speaking of lenny, those US DID's are dead. Does anyone have a US DID > they can point? > > On Jan 8, 2013, at 12:27 AM, jay binks wrote: > > So what would i have to do to win that :) > ( yea I setup lenny, and kinda look after him when I can find time ) > > I kinda knew about it but discounted it since im not in the USA. > > On 6 January 2013 19:04, Gabriel Gunderson wrote: > >> Will someone please forward this valuable infomation to Lenny? >> >> http://money.cnn.com/2013/01/02/pf/ftc-robocalls/index.html > > > > ---------- Forwarded message ---------- > From: Michael Jerris > To: FreeSWITCH Users Help > Cc: > Date: Tue, 8 Jan 2013 08:35:53 -0500 > Subject: Re: [Freeswitch-users] Question on Sevens Erlang Page > or if you want it in english: > > > http://translate.google.co.ve/translate?hl=en&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search%3Fq%3Dbuild%2Ba%2Bcomplex%2Bhence%2Bpowerful%2Bfreeswitch%2Bivr%2Bin%2Berlang%26hl%3Des-419%26client%3Dfirefox-a%26hs%3D06u%26tbo%3Dd%26rls%3Dorg.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg > > > On Jan 8, 2013, at 8:25 AM, Saugort Dario Garcia Tovar < > dgarcia at anew.com.ve> wrote: > > Hi, > > Check this link: > > http://translate.google.co.ve/translate?hl=es-419&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search%3Fq%3Dbuild%2Ba%2Bcomplex%2Bhence%2Bpowerful%2Bfreeswitch%2Bivr%2Bin%2Berlang%26hl%3Des-419%26client%3Dfirefox-a%26hs%3D06u%26tbo%3Dd%26rls%3Dorg.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg > > Original link > > http://www.google.co.ve/url?sa=t&rct=j&q=&esrc=s&source=web&cd=4&ved=0CEEQFjAD&url=http%3A%2F%2Fwww.freeswitch.org.cn%2F2010%2F04%2F26%2Fshi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&ei=CxzsUKu5Oo-K9QTh_ICAAg&usg=AFQjCNFxsI4KHYigeKt2ZVQsPkSrwwiW5g&bvm=bv.1357316858,d.eWU&cad=rja > > I dont know if the link has the same info. > > > On 1/8/2013 4:48 AM, Gerald Weber wrote: > > Hi,**** > > ** ** > > i was looking fort he page**** > > > http://www.dujinfang.com/past/2010/4/22/build-a-complex-hence-powerful-freeswitch-ivr-in-erlang > **** > > ** ** > > The page seems to be offline for maintenance for weeks now and i?m not > able to find it either in googles cache or wayback.archive.org.**** > > ** ** > > does anyone have a copy of hat page (seven maybe?) **** > > ** ** > > thanks in advance,**** > > gw**** > > ** ** > > Ps: what happened to this page ?**** > > ** ** > > > > ---------- Forwarded message ---------- > From: Seven Du > To: FreeSWITCH Users Help > Cc: > Date: Tue, 8 Jan 2013 21:50:46 +0800 > Subject: Re: [Freeswitch-users] Question on Sevens Erlang Page > The site was hosted on heroku but when heroku upgraded it's ruby stack it > made me hard to upgrade my blog software and it was offline for a couple > weeks because I was too busy to update it. > > I move all to github pages now so here's the new link: > > > http://www.dujinfang.com/2010/04/22/build-a-complex-hence-powerful-freeswitch-ivr-in-erlang.html > > Thanks all. > > -- > Seven Du > Sent with Sparrow > > On Tuesday, January 8, 2013 at 9:35 PM, Michael Jerris wrote: > > or if you want it in english: > > > http://translate.google.co.ve/translate?hl=en&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search%3Fq%3Dbuild%2Ba%2Bcomplex%2Bhence%2Bpowerful%2Bfreeswitch%2Bivr%2Bin%2Berlang%26hl%3Des-419%26client%3Dfirefox-a%26hs%3D06u%26tbo%3Dd%26rls%3Dorg.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg > > > On Jan 8, 2013, at 8:25 AM, Saugort Dario Garcia Tovar < > dgarcia at anew.com.ve> wrote: > > Hi, > > Check this link: > > http://translate.google.co.ve/translate?hl=es-419&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search%3Fq%3Dbuild%2Ba%2Bcomplex%2Bhence%2Bpowerful%2Bfreeswitch%2Bivr%2Bin%2Berlang%26hl%3Des-419%26client%3Dfirefox-a%26hs%3D06u%26tbo%3Dd%26rls%3Dorg.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg > > Original link > > http://www.google.co.ve/url?sa=t&rct=j&q=&esrc=s&source=web&cd=4&ved=0CEEQFjAD&url=http%3A%2F%2Fwww.freeswitch.org.cn%2F2010%2F04%2F26%2Fshi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&ei=CxzsUKu5Oo-K9QTh_ICAAg&usg=AFQjCNFxsI4KHYigeKt2ZVQsPkSrwwiW5g&bvm=bv.1357316858,d.eWU&cad=rja > > I dont know if the link has the same info. > > > On 1/8/2013 4:48 AM, Gerald Weber wrote: > > Hi,**** > > ** ** > > i was looking fort he page**** > > > http://www.dujinfang.com/past/2010/4/22/build-a-complex-hence-powerful-freeswitch-ivr-in-erlang > **** > > ** ** > > The page seems to be offline for maintenance for weeks now and i?m not > able to find it either in googles cache or wayback.archive.org.**** > > ** ** > > does anyone have a copy of hat page (seven maybe?) **** > > ** ** > > thanks in advance,**** > > gw**** > > ** ** > > Ps: what happened to this page ?**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/ec92896e/attachment-0001.html From benoit.raymond at amiconcept.com Wed Jan 9 06:15:07 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Tue, 8 Jan 2013 22:15:07 -0500 Subject: [Freeswitch-users] mod_spandsp keeps returning "error reading frame" Message-ID: <033001cdee17$86e8e1c0$94baa540$@amiconcept.com> Hi, I've been playing with spandsp for a couple of days already and couldn't find much discussion about it to make it work. First, I have tried it from the dialplan: Then I thought it was due to the fact the call hasn't been answered yet so reverted to create a lua script and test again. This is the lua code I used: originate_base = "{ignore_early_media=true,originate_timeout=90,hangup_after_bridge=true,leg= 1}"; dialstr = "sofia/gateway/mygw/1234"; mySession = null; ostr = originate_base..dialstr; con = freeswitch.EventConsumer("DETECTED_TONE"); freeswitch.consoleLog("info", "Dialing Leg: " .. ostr .. "\n"); mySession = freeswitch.Session(ostr); local hcause = mySession:hangupCause(); if ( mySession:ready() ) then freeswitch.consoleLog("info", "We are connected. Start tone detection\n"); mySession:execute("sleep", 250); mySession:execute("start_tone_detect", "alarm_panel"); -- Wait for the HANDSHAKE freeswitch.consoleLog("info", "Now we wait for the HANDSHAKE\n"); mySession:execute("sleep", 5000); freeswitch.consoleLog("info", "After the 5 seconds sleep\n"); for e in (function() return con:pop(1,2000) end) do freeswitch.consoleLog("info", "Got an Event: " .. e:serialize("xml") .. "\n"); end -- I guess we are done mySession:hangup(); end Nada . while the call is answered already, it is still failing: 2013-01-08 21:40:13.835554 [NOTICE] switch_cpp.cpp:78 bound to DETECTED_TONE 2013-01-08 21:40:13.835554 [INFO] switch_cpp.cpp:1227 Dialing Leg: {ignore_early_media=true,originate_timeout=90,hangup_after_bridge=true,leg=1 ,uuid=9876543210}sofia/gateway/mygw/1234 2013-01-08 21:40:13.835554 [NOTICE] switch_channel.c:926 New Channel sofia/external/1234 [e50a57ac-5a05-11e2-9f6a-996996505a23] 2013-01-08 21:40:14.315558 [NOTICE] sofia.c:6428 Channel [sofia/external/1234] has been answered 2013-01-08 21:40:14.335571 [INFO] switch_cpp.cpp:1227 We are connected. Start tone detection 2013-01-08 21:40:14.595560 [INFO] mod_spandsp_dsp.c:464 (sofia/external/1234) initializing tone detector 2013-01-08 21:40:14.595560 [INFO] switch_cpp.cpp:1227 Now we wait for the HANDSHAKE 2013-01-08 21:40:14.615556 [INFO] mod_spandsp_dsp.c:475 (sofia/external/1234) error reading frame 2013-01-08 21:40:14.615556 [INFO] mod_spandsp_dsp.c:500 (sofia/external/1234) destroying tone detector 2013-01-08 21:40:19.615559 [INFO] switch_cpp.cpp:1227 After the 5 seconds sleep 2013-01-08 21:40:21.615561 [NOTICE] switch_cpp.cpp:637 Hangup sofia/external/1234 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] The spandsp.conf.xml: It is loading as per the DEBUG output: 2013-01-08 22:08:39.900772 [DEBUG] mod_spandsp.c:449 Adding tone_descriptor: alarm_panel 2013-01-08 22:08:39.900772 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: alarm_panel, tone: HANDSHAKE(0) 2013-01-08 22:08:39.900772 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: alarm_panel, tone: HANDSHAKE(0), element (1400, 0, 95, 105) 2013-01-08 22:08:39.900772 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: alarm_panel, tone: HANDSHAKE(0), element (0, 0, 95, 105) 2013-01-08 22:08:39.900772 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: alarm_panel, tone: HANDSHAKE(0), element (2300, 0, 95, 105) 2013-01-08 22:08:39.900772 [DEBUG] mod_spandsp.c:470 Adding tone_descriptor: alarm_panel, tone: KISSOFF(1) 2013-01-08 22:08:39.900772 [DEBUG] mod_spandsp.c:505 Adding tone_descriptor: alarm_panel, tone: KISSOFF(1), element (1400, 0, 750, 1000) 2013-01-08 22:08:39.900772 [DEBUG] mod_spandsp_fax.c:192 FAX timer thread started. 2013-01-08 22:08:39.900772 [DEBUG] mod_spandsp.c:574 mod_spandsp loaded, using spandsp library version [20100724 163333] 2013-01-08 22:08:39.900772 [CONSOLE] switch_loadable_module.c:1318 Successfully Loaded [mod_spandsp] I am using freeswitch 1.2: freeswitch at internal> version FreeSWITCH Version 1.2.0 (git-1eae279 2012-05-14 13-07-40 -0500) I tried using the tone_detect from Freeswitch core but it is not consistent . Anyone made mod_spandsp working? Thanks, Benoit Raymond AMI Concept Inc. Business VoIP Solutions / Web Design Phone: (450) 553-1231 http://www.amiconcept.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130108/89a38338/attachment.html From gerald.weber at besharp.at Wed Jan 9 09:35:27 2013 From: gerald.weber at besharp.at (Gerald Weber) Date: Wed, 9 Jan 2013 06:35:27 +0000 Subject: [Freeswitch-users] Question on Sevens Erlang Page In-Reply-To: References: <50EC1E47.5050409@anew.com.ve> <2B6D1002-DD1A-4F20-8128-AA80C940D29B@jerris.com> Message-ID: Thanks all ! Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Seven Du Gesendet: Dienstag, 08. J?nner 2013 14:51 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Question on Sevens Erlang Page The site was hosted on heroku but when heroku upgraded it's ruby stack it made me hard to upgrade my blog software and it was offline for a couple weeks because I was too busy to update it. I move all to github pages now so here's the new link: http://www.dujinfang.com/2010/04/22/build-a-complex-hence-powerful-freeswitch-ivr-in-erlang.html Thanks all. -- Seven Du Sent with Sparrow On Tuesday, January 8, 2013 at 9:35 PM, Michael Jerris wrote: or if you want it in english: http://translate.google.co.ve/translate?hl=en&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search%3Fq%3Dbuild%2Ba%2Bcomplex%2Bhence%2Bpowerful%2Bfreeswitch%2Bivr%2Bin%2Berlang%26hl%3Des-419%26client%3Dfirefox-a%26hs%3D06u%26tbo%3Dd%26rls%3Dorg.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg On Jan 8, 2013, at 8:25 AM, Saugort Dario Garcia Tovar > wrote: Hi, Check this link: http://translate.google.co.ve/translate?hl=es-419&sl=zh-CN&u=http://www.freeswitch.org.cn/2010/04/26/shi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&prev=/search%3Fq%3Dbuild%2Ba%2Bcomplex%2Bhence%2Bpowerful%2Bfreeswitch%2Bivr%2Bin%2Berlang%26hl%3Des-419%26client%3Dfirefox-a%26hs%3D06u%26tbo%3Dd%26rls%3Dorg.mozilla:en-US:official&sa=X&ei=9RzsUKC1E4rQ9AT86ICQCQ&ved=0CEoQ7gEwAg Original link http://www.google.co.ve/url?sa=t&rct=j&q=&esrc=s&source=web&cd=4&ved=0CEEQFjAD&url=http%3A%2F%2Fwww.freeswitch.org.cn%2F2010%2F04%2F26%2Fshi-yong-erlangjian-li-ivrshi-xian-fu-za-ye-wu-luo-ji.html&ei=CxzsUKu5Oo-K9QTh_ICAAg&usg=AFQjCNFxsI4KHYigeKt2ZVQsPkSrwwiW5g&bvm=bv.1357316858,d.eWU&cad=rja I dont know if the link has the same info. On 1/8/2013 4:48 AM, Gerald Weber wrote: Hi, i was looking fort he page http://www.dujinfang.com/past/2010/4/22/build-a-complex-hence-powerful-freeswitch-ivr-in-erlang The page seems to be offline for maintenance for weeks now and i?m not able to find it either in googles cache or wayback.archive.org. does anyone have a copy of hat page (seven maybe?) thanks in advance, gw Ps: what happened to this page ? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/ebc54a89/attachment-0001.html From oej at edvina.net Wed Jan 9 11:29:50 2013 From: oej at edvina.net (Olle E. Johansson) Date: Wed, 9 Jan 2013 09:29:50 +0100 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: References: Message-ID: <2D95B81A-2356-43BE-9AC0-A2D0BDC741AB@edvina.net> 8 jan 2013 kl. 21:46 skrev Kristian Kielhofner : > The correct positive response to an UPDATE would be a 200 that is then > ACKd. See RFC 3311: The only transaction that needs an ACK is INVITE. RFC3311 does not mention an ACK as a response to UPDATE. /O > > http://tools.ietf.org/html/rfc3311 > > On Tue, Jan 8, 2013 at 2:40 PM, Steven Ayre wrote: >> Does the client send an ACK for the UPDATE? >> > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From michel.brabants at gmail.com Wed Jan 9 12:48:10 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Wed, 9 Jan 2013 10:48:10 +0100 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: References: Message-ID: No, freeswitch is not getting a response as the client (B2B sipserver) converts this to a reinvite with no sdp (which is incorrect of course). However, if I need to wait in the client-developers to fix this, I'll be a half year further... I was wondering however why freeswitch is sending these in the first place as it contains no differences from previous sdp's. It isn't also sending updates or reinvites to the other side. I don't see the purpose of these updates. If it was to change the code, ... it would be better in this case as freeswitch could send reinvites instead as these wouldn't be handled incorrectly normally by the client. Michel On Tue, Jan 8, 2013 at 10:11 PM, Steven Ayre wrote: > Yep my mistake... rather I meant is FS getting a response? > (Wondering if FS is retransmitting and/or timing out the call on no > response since you say it hangs up 30s later) > > > > On 8 January 2013 20:46, Kristian Kielhofner wrote: > > The correct positive response to an UPDATE would be a 200 that is then > > ACKd. See RFC 3311: > > > > http://tools.ietf.org/html/rfc3311 > > > > On Tue, Jan 8, 2013 at 2:40 PM, Steven Ayre wrote: > >> Does the client send an ACK for the UPDATE? > >> > > > > > > -- > > Kristian Kielhofner > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/405d687b/attachment.html From Alexander.Haugg at c4b.de Wed Jan 9 15:03:41 2013 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Wed, 9 Jan 2013 12:03:41 +0000 Subject: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project Message-ID: Hi, the project libmpg123 do not build in debug mode. The error is in the "Download mpg123.2012.vcxproj". see the fix in my git diff: diff --git libs/win32/Download mpg123.2012.vcxproj libs/win32/Download mpg123.2012.vcxproj index ead3f9b..9942004 100644 --- libs/win32/Download mpg123.2012.vcxproj +++ libs/win32/Download mpg123.2012.vcxproj @@ -68,7 +68,9 @@ Document Downloading mpg123. - if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." + if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." +if not exist "$(ProjectDir)..\libmpg123" move "$(ProjectDir)..\mpg123-1.14.4" "$(ProjectDir)..\libmpg123" + $(ProjectDir)..\libmpg123;%(Outputs) Downloading mpg123. if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." In the debug mode the mpg123-1.14.4 folder will not renamend to libmpg123. Is it possible to fix it? Thanks for your help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/59b00556/attachment.html From kevin.jzh at hotmail.com Wed Jan 9 15:47:11 2013 From: kevin.jzh at hotmail.com (KevinJin) Date: Wed, 9 Jan 2013 20:47:11 +0800 Subject: [Freeswitch-users] How to configure the freeswitch.dbh ODBC log Message-ID: HI All, my Lua script using the freeswitch.dbh (ODBC) to connect to the postgresql and it works, however I find that freeswitch creates odbc connecting logs in different folders, I find two logs in the conf/autoload_configs/ 16600.log, 16603.log, and one log in the freeswitch base folder /opt/freeswitch/ 12324.log. It's really strang. How can I configure the system to put these logs in one place? and how to turn off these logs? =============snip =========================[3061185392]calling getDSNdefaults conn = 0xb713ade8, PGAPI_Connect(DSN='cb', UID='conf', PWD='xxxxx') [3061185392]CC_connect: entering... [3061185392]sslmode=disable [3061185392]original_CC_connect: entering... Driver Version='08.04.0200,200912260001' [3061185392]Driver Version='08.04.0200,200912260001' Global Options: fetch=100, socket=4096, unknown_sizes=0, max_varchar_size=255, max_longvarchar_size=65536 disable_optimizer=0, ksqo=1, unique_index=1, use_declarefetch=0 text_as_longvarchar=1, unknowns_as_longvarchar=0, bools_as_char=1 NAMEDATALEN=64 extra_systable_prefixes='dd_;', conn_settings='' conn_encoding='' [3061185392]original_CC_connect: DSN = 'cb', server = 'localhost', port = '5432', database = 'cb', username = 'conf', password='xxxxx' [3061185392]connecting to the server socket... [3061185392](0)(null) ERRNO=115 [3061185392]connection to the server socket succeeded. [3061185392]sizeof startup packet = 292 [3061185392]sent the authentication block successfully. [3061185392]gonna do authentication [3061185392]read -1, global_socket_buffersize=4096 [3061185392]Lasterror=11 [3061185392]read 15, global_socket_buffersize=4096 [3061185392]auth got 'R' ----snip ------[3060202352][SQLGetData][3060202352]PGAPI_GetData: enter, stmt=0xb7145ad0 icol=1 [3060202352] num_rows = 1 [3060202352] value = '1000' [3060202352]**** PGAPI_GetData: icol = 0, target_type = 1, field_type = 23, value = '1000' [3060202352]copy_and_convert: field_type = 23, fctype = 1, value = '1000', cbValueMax=11 [3060202352]DEFAULT: len = 4, ptr = '1000' [3060202352] SQL_C_CHAR, default: len = 4, cbValueMax = 11, rgbValueBindRow = '1000' [3060202352][SQLDescribeCol][3060202352]PGAPI_DescribeCol: entering.2.. [3060202352]SC_pre_execute: status = 3 [3060202352]PGAPI_DescribeCol: result = 0xb7140688, status = 3, numcols = 3 [3060202352]describeCol: col 1 fieldname = 'conf_pin' [3060202352]describeCol: col 1 fieldtype = 23 [3060202352]describeCol: col 1 column_size = 10 [3060202352]describeCol: col 1 *pfSqlType = 4 [3060202352]describeCol: col 1 *pcbColDef = 10 [3060202352]describeCol: col 1 *pibScale = 0 [3060202352]describeCol: col 1 *pfNullable = 1 [3060202352][SQLGetData][3060202352]PGAPI_GetData: enter, stmt=0xb7145ad0 icol=2 [3060202352] num_rows = 1 [3060202352] value = '1234'----snip ------- Thanks,Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/e74ae71e/attachment-0001.html From jmesquita at freeswitch.org Wed Jan 9 16:07:09 2013 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 9 Jan 2013 10:07:09 -0300 Subject: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project In-Reply-To: References: Message-ID: <0E9365BE-1112-4040-9C6F-D237E14B657A@freeswitch.org> Dude, JIRA!! We will say that as many times as we need to aren't we? Sent from my iPhone On Jan 9, 2013, at 9:03 AM, Alexander Haugg wrote: > Hi, > > the project libmpg123 do not build in debug mode. The error is in the ?Download mpg123.2012.vcxproj?. > see the fix in my git diff: > > diff --git libs/win32/Download mpg123.2012.vcxproj libs/win32/Download mpg123.2012.vcxproj > index ead3f9b..9942004 100644 > --- libs/win32/Download mpg123.2012.vcxproj > +++ libs/win32/Download mpg123.2012.vcxproj > @@ -68,7 +68,9 @@ > > Document > Downloading mpg123. > - if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." > + if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." > +if not exist "$(ProjectDir)..\libmpg123" move "$(ProjectDir)..\mpg123-1.14.4" "$(ProjectDir)..\libmpg123" > + > $(ProjectDir)..\libmpg123;%(Outputs) > Downloading mpg123. > if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." > > In the debug mode the mpg123-1.14.4 folder will not renamend to libmpg123. > > Is it possible to fix it? > > Thanks for your help. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/0602d965/attachment.html From oej at edvina.net Wed Jan 9 11:47:02 2013 From: oej at edvina.net (Olle E. Johansson) Date: Wed, 9 Jan 2013 09:47:02 +0100 Subject: [Freeswitch-users] Best practices question about SIP registration * IPv6 lookups In-Reply-To: References: <50DE75C4.1030500@communicatefreely.net> Message-ID: From the wiki: "There IS a wrinkle to this special case: If you have a URL of the form the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host example.com (i" This is an error in the RFC that we are trying to fix in the IPV6 wg of the SIP Forum. Software should NOT find A *OR* AAAA. It should look up both and then try them according to Happy Eyeballs - in parallell with a small advantage for IPv6. If we do serial forking here, we have in worst case a 32 second timeout in SIP over UDP, which is not acceptable. Every customer should require fixes and every developer should try to fix this... Some slides on these issues: * http://www.slideshare.net/oej/sipforum-sip-ipv6-discussion-slides * http://www.slideshare.net/oej/sip-ipv6-time-for-action I hope to see FreeSwitch at SIPit 30 in February so we can test this and other issues! /O 9 jan 2013 kl. 00:21 skrev Michael Collins : > I couldn't stand it, so I added Lawrence's info to this page: > http://wiki.freeswitch.org/wiki/DNS#Explanation_of_DNS.2FSRV.2FNAPTR > > Cheers, > MC > > On Tue, Jan 8, 2013 at 2:56 PM, Michael Collins wrote: > Lawrence, > > Thanks for this explanation. It was very well written. I'm looking for a place to codify this on the wiki so that it gets preserved... :) > > -MC > > > On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy wrote: > Hi there, > at the risk of butting in on someone else's party ... > Nope; your interpretations is NOT best practice. > I have some sympathy, as the term domain is overloaded within fS. > > A sip address consists of a userpart and a domain part -- e.g., > The sip domain is similar to an email domain -- e.g., > With email, you need to do a lookup of the MX record in DNS to find the FQDN of the machine that handles mail for the domain. > With SIP (see RFC 3263), you do a lookup on the SRV record (at _sip._udp.) to find the machine that handles SIP > registrations/incalls for the domain. That also gives you the port on which that machine is listening. > (Yup, you can also have a NAPTR record in the domain to tell you where the SRV record is, but many folks don't bother -- for Best Practice, you should, but ...) > > There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) support that means most SIP clients will look for the SRV and, if it can't be found (or there's an IP address rather than a DNS -style domain, in which case the SIP client won't bother hunting the SRV), the client will guess that the domain has a machine (i.e. it will look for an A or AAAA record), and also guess it's listening on 5060 (the default port). > Email is the same (mail to fred at example.com, and strictly the sender will do a check for a MX and then look for an A record for example.com, and try there). > > However, relying on that default "get out" clause is definitely NOT what you should do for BCP. > Using the hostname as the sip domain is a kludge -- the FQDN with A record usually works, but it's not what you want to do. > > SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at _sip._udp., and you're done. No need for an A record at that domain at all. > > (RFC 3263 is not too hard to read, for a change -- it's certainly shorter than RFC 3261, and it even has an ASCII art diagram :). > > all the best, > Lawrence > > On 8 Jan 2013, at 21:05, Steven Schoch wrote: > > > On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < > > fs-list at communicatefreely.net> wrote: > > > >> Hi Steven, > >> > >> I would recommend using a proper domain name as much as possible. For > >> one, it looks > >> nicer! A SIP URI is supposed to be user at domain like an e-mail address > >> is, and I hope that > >> one day URI dialing will be common place, so we might as well do it right > >> the first time. > >> > > > > What you're saying is that "domain" should really be a fully-qualified host > > name that points via DNS to the actual host on which FreeSwitch is running. > > That is, the domain should be "pbx.example.com" instead of just " > > example.com", as the last example would most likely point to a web server, > > not the SIP server. Do I have that right? > > > > Next, in the configuration for Polycom phones (for example), there are 2 > > fields that both have the userid. In the example in > > http://wiki.freeswitch.org/wiki/Polycom_configuration it has: > > > > reg.1.auth.userId="1000" > > > > and > > > > reg.1.address="1000 at fs.domain.local" > > > > How is the "address" value used? Is that sent in the SIP registration > > message? If that's the case, what does Freeswitch do with it? > > > > -- > > Steve > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/74b2c0c2/attachment-0001.html From mike at jerris.com Wed Jan 9 16:21:58 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Jan 2013 08:21:58 -0500 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: References: Message-ID: <1871B0BF-023F-4665-8398-0626556FA72F@jerris.com> Its to update the caller id information. you can disable this: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#send-display-update Mike On Jan 9, 2013, at 4:48 AM, Michel Brabants wrote: > No, freeswitch is not getting a response as the client (B2B sipserver) converts this to a reinvite with no sdp (which is incorrect of course). However, if I need to wait in the client-developers to fix this, I'll be a half year further... > > I was wondering however why freeswitch is sending these in the first place as it contains no differences from previous sdp's. It isn't also sending updates or reinvites to the other side. I don't see the purpose of these updates. If it was to change the code, ... it would be better in this case as freeswitch could send reinvites instead as these wouldn't be handled incorrectly normally by the client. > > > Michel > > On Tue, Jan 8, 2013 at 10:11 PM, Steven Ayre wrote: > Yep my mistake... rather I meant is FS getting a response? > (Wondering if FS is retransmitting and/or timing out the call on no > response since you say it hangs up 30s later) > > > > On 8 January 2013 20:46, Kristian Kielhofner wrote: > > The correct positive response to an UPDATE would be a 200 that is then > > ACKd. See RFC 3311: > > > > http://tools.ietf.org/html/rfc3311 > > > > On Tue, Jan 8, 2013 at 2:40 PM, Steven Ayre wrote: > >> Does the client send an ACK for the UPDATE? > >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/c7c1d440/attachment.html From peter.olsson at visionutveckling.se Wed Jan 9 16:26:20 2013 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 9 Jan 2013 13:26:20 +0000 Subject: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project Message-ID: <1FFF97C269757C458224B7C895F35F151FBC15@cantor.std.visionutv.se> This is already being handled in this Jira: FS-4768 (together with other changes for VS2012). /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Jo?o Mesquita Skickat: den 9 januari 2013 14:07 Till: FreeSWITCH Users Help Kopia: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project Dude, JIRA!! We will say that as many times as we need to aren't we? Sent from my iPhone On Jan 9, 2013, at 9:03 AM, Alexander Haugg > wrote: Hi, the project libmpg123 do not build in debug mode. The error is in the ?Download mpg123.2012.vcxproj?. see the fix in my git diff: diff --git libs/win32/Download mpg123.2012.vcxproj libs/win32/Download mpg123.2012.vcxproj index ead3f9b..9942004 100644 --- libs/win32/Download mpg123.2012.vcxproj +++ libs/win32/Download mpg123.2012.vcxproj @@ -68,7 +68,9 @@ Document Downloading mpg123. - if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." + if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." +if not exist "$(ProjectDir)..\libmpg123" move "$(ProjectDir)..\mpg123-1.14.4" "$(ProjectDir)..\libmpg123" + $(ProjectDir)..\libmpg123;%(Outputs) Downloading mpg123. if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." In the debug mode the mpg123-1.14.4 folder will not renamend to libmpg123. Is it possible to fix it? Thanks for your help. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:50ed68b632765624898328! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/d64db695/attachment.html From Alexander.Haugg at c4b.de Wed Jan 9 17:04:07 2013 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Wed, 9 Jan 2013 14:04:07 +0000 Subject: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project In-Reply-To: <1FFF97C269757C458224B7C895F35F151FBC15@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F151FBC15@cantor.std.visionutv.se> Message-ID: Thanks Alex Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Peter Olsson Gesendet: Mittwoch, 9. Januar 2013 14:26 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project This is already being handled in this Jira: FS-4768 (together with other changes for VS2012). /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Jo?o Mesquita Skickat: den 9 januari 2013 14:07 Till: FreeSWITCH Users Help Kopia: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project Dude, JIRA!! We will say that as many times as we need to aren't we? Sent from my iPhone On Jan 9, 2013, at 9:03 AM, Alexander Haugg > wrote: Hi, the project libmpg123 do not build in debug mode. The error is in the ?Download mpg123.2012.vcxproj?. see the fix in my git diff: diff --git libs/win32/Download mpg123.2012.vcxproj libs/win32/Download mpg123.2012.vcxproj index ead3f9b..9942004 100644 --- libs/win32/Download mpg123.2012.vcxproj +++ libs/win32/Download mpg123.2012.vcxproj @@ -68,7 +68,9 @@ Document Downloading mpg123. - if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." + if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." +if not exist "$(ProjectDir)..\libmpg123" move "$(ProjectDir)..\mpg123-1.14.4" "$(ProjectDir)..\libmpg123" + $(ProjectDir)..\libmpg123;%(Outputs) Downloading mpg123. if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." In the debug mode the mpg123-1.14.4 folder will not renamend to libmpg123. Is it possible to fix it? Thanks for your help. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:50ed68b632765624898328! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/769d5715/attachment-0001.html From fs-list at communicatefreely.net Wed Jan 9 18:36:35 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Wed, 09 Jan 2013 10:36:35 -0500 Subject: [Freeswitch-users] SIP packet exceeds MTU? Message-ID: <50ED8E83.3030707@communicatefreely.net> Hello, I'm having a problem with BLF in some cases, and I found that the length of the username was a factor in whether the lamps worked or not. When I did some packet captures, I found that problem packets were 1500 bytes or so. Since the endpoint is on a DSL line that has an MTU of 1460, the packet doesn't fit. I'm already using compressed headers. Is there a way to reduce the packet size more, or is there another solution? I will include an example below: Note that my IP address and domain substutions may change the packet size a little, but the point is the same. Thanks! NOTIFY sip:XXXX at YYY.YYY.YYY.YYY:5060;transport=udp SIP/2.0 v:SIP/2.0/UDP zzz.zzz.zzz.zzz:5070;rport;branch=z9hG4bKD5QUr5m2Fm3Dp Route:;transport=udp Max-Forwards:70 f:;tag=TrmHE4FBIrB4 t:"Tim St. Pierre";tag=ecc2498bfd i:3decd12decccafe1 CSeq:448376002 NOTIFY m: User-Agent:Communicate Freely 2.0 Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE k:timer,precondition,path,replaces o:dialog u:talk,hold,conference,presence,dialog,line-seize,call-info,sla,include-session-description,presence.winfo,message-summary,refer Subscription-State:active;expires=1137 c:application/dialog-info+xml l:639 confirmed sip:accountcode-701 at ourdomain.com;proto=park sip:accountcode-701 References: <50ED8E83.3030707@communicatefreely.net> Message-ID: Unfortunately if you're using compressed headers then the packet's pretty much as small as it can get. Can you use TCP as the transport for SIP? If so that should handle the fragmentation ok. -Steve On 9 January 2013 15:36, Tim St. Pierre wrote: > I'm already using compressed headers. Is there a way to reduce the packet size more, or > is there another solution? I will include an example below: Note that my IP address and > domain substutions may change the packet size a little, but the point is the same. From sdevoy at bizfocused.com Wed Jan 9 18:57:32 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 9 Jan 2013 10:57:32 -0500 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: <1ADA846F-0D19-4FE3-AFA3-7CA5E8E4C127@insensate.co.uk> References: <50DE75C4.1030500@communicatefreely.net> <1ADA846F-0D19-4FE3-AFA3-7CA5E8E4C127@insensate.co.uk> Message-ID: <2ef401cdee82$090f9df0$1b2ed9d0$@bizfocused.com> Great info Lawrence, especially the MX analogy. Thanks. I have to take this one more step and ask a question. We have FS configured for multitenant. Of course the "domains" for the tenants are FQDNs. I found it simple to setup and wonderful for setting up customers as their own virtual switches. Clearly this is a case where the sip address has to contain the FQDN. (It also gives me a chuckle every day when I see those a-holes' registration failures for "100@" when a name is required!) My FS Domains are like "fs_A.xyz.com" and "FS_B.xyz.com" which I currently have as A records with the same IP address. If I add an SRV record for _sip._udp.xyz.com to that IP Address, are you saying I won't need the A records? Or do I need SRV records for each sub-domain? Of course this is assuming I set my hardware (Cisco SPA 5xx) to "Use SRV? Yes" Thanks for the information. Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lawrence Conroy Sent: Tuesday, January 08, 2013 7:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best practices question about SIP registration Hi again Steven, with apologies for top-posting ... Re. flowroute.com Many of us have sup-domains for different classes of users -- looks like your sub-domain is sip.flowroute.com. However, flowroute.com does have a NAPTR (it has two): ;; ANSWER SECTION: flowroute.com. 21600 IN NAPTR 100 10 "s" "SIP+D2U" "" _sip._udp.flowroute.com. flowroute.com. 21600 IN NAPTR 102 20 "s" "SIP+D2T" "" _sip._tcp.flowroute.com. Also, the _sip._udp.flowroute SRV shows: ;; ANSWER SECTION: _sip._udp.flowroute.com. 43200 IN SRV 20 10 5060 sip-ca1.flowroute.com. _sip._udp.flowroute.com. 43200 IN SRV 10 10 5060 sip-nv1.flowroute.com. as does the _sip.tcp.flowroute.com SRV: ;; ANSWER SECTION: _sip._tcp.flowroute.com. 43200 IN SRV 10 10 5060 sip-nv1.flowroute.com. _sip._tcp.flowroute.com. 43200 IN SRV 20 10 5060 sip-ca1.flowroute.com. so -- if a client is trying to call to sip:1000 at flowroute.com, that client looks up the NAPTR set for flowroute.com (the sip domainpart), gets back the two entries, and chooses to look for a SRV under _sip._udp.flowroute.com (if it wants to call using UDP) or _sip.tcp.flowroute.com (if it wants to call using TCP or use sips:). flowroute would prefer that you contacted them by UDP (it has a lower preference -- 10 versus the D2T's 20). If you do contact them, they'd prefer that you used sip-nv1.flowroute.com. (as that has a better/lower preference than sip-cal). On to the polycoms -- I don't have any of these, so I'm guessing, but ... It looks strongly like the registration server is at fs.domain.local, which has an IP address of 10.10.10.11 for the DNS-challenged, and is listening on port 5060. The AuthID/UserID for this 'phone to use 1000, which is kinda apparent from the SIP address the phone has, which is sip:1000 at fs.domain.local. I'd AssUMe that there's also a password field :). So ... if you had mydomain.com, and for your internal use you had a sub-domain of, say, internal.mydomain.com, and you had one fS handling your local 'phones, running on a machine called fs.internal.domain.com which had an IP address of 10.10.10.11, I'd put into the local view of DNS: internal.mydomain.com. IN NAPTR 100 10 "s" "SIP+D2U" "" _sip._udp.internal.mydomain.com. _sip._udp.internal.mydomain.com. IN SRV 10 10 5060 fs.internal.mydomain.com fs.internal.mydomain.com. IN A 10.10.10.11 Note that I'm assuming that you'd be using a split-view DNS (i.e., your local DNS server would only give these answers to its local machines) as it's kinda impolite to give private addresses like 10.10.10.11 to the outside world -- clients are sometimes too dumb to realise they can't contact that server using this address. [...which is why we have one way audio -- stupid implementers :]p Hope that helps, Lawrence On 8 Jan 2013, at 23:02, Steven Schoch wrote: > On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy wrote: > >> Hi there, >> at the risk of butting in on someone else's party ... >> Nope; your interpretations is NOT best practice. >> > > This party is open to all. I appreciate these answers, and as usual I > have noticed that the fastest way to learn is to do it wrong, and then > have somebody correct me! Thanks! > > >> With SIP (see RFC 3263), you do a lookup on the SRV record (at >> _sip._udp.) to find the machine that handles SIP >> registrations/incalls for the domain. That also gives you the port on >> which that machine is listening. >> > > For example, I did a lookup: > > $ dig SRV _sip._udp.sip.flowroute.com. > > This returns: > > _sip._udp.sip.flowroute.com. 43200 IN SRV 20 10 5060 > sip-ca1.flowroute.com. > _sip._udp.sip.flowroute.com. 43200 IN SRV 10 10 5060 > sip-nv1.flowroute.com. > > > >> (Yup, you can also have a NAPTR record in the domain to tell you >> where the SRV record is, but many folks don't bother -- for Best >> Practice, you should, but ...) >> > > $ dig NAPTR sip.flowroute.com > > sip.flowroute.com. 42818 IN NAPTR 100 10 "s" "SIP+D2U" "" > _sip._udp.sip.flowroute.com. > sip.flowroute.com. 42818 IN NAPTR 102 20 "s" "SIP+D2U" "" > _sip._tcp.sip.flowroute.com. > > I found that _sip._tcp.sip.flowroute.com does not have a SRV record. > I don't know what that means. The RFC says that you should use > "SIP+D2T" for TCP. > > Now, as this relates to the Polycom phone example, there are these fields: > > reg.1.address="1000 at fs.domain.local" > reg.1.auth.userId="1000" > reg.1.server.1.address="10.10.10.11" > reg.1.server.1.port="5060" > > Since this is a VoIP phone that loads its configuration from the > server, not VoIP software that is configured by the user, it really > only needs the IP address, so the server address field is kind of moot. > > But the example domain here is "fs.domain.local". Why not just > "domain.local"? Why didn't Flowroute use "flowroute.com" instead of " > sip.flowroute.com"? > > And my other question was how is the address "1000 at fs.domain.local" used? > Does FreeSwitch use that or does it just use the auth.userId? > > -- > Steve > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From bdfoster at endigotech.com Wed Jan 9 18:59:52 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 9 Jan 2013 10:59:52 -0500 Subject: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project In-Reply-To: References: <1FFF97C269757C458224B7C895F35F151FBC15@cantor.std.visionutv.se> Message-ID: <096B0F95-7A32-422E-B5B8-66CA83B47A80@endigotech.com> Sent from my iPhone On Jan 9, 2013, at 9:04 AM, Alexander Haugg wrote: > Thanks > > Alex > > Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Peter Olsson > Gesendet: Mittwoch, 9. Januar 2013 14:26 > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project > > This is already being handled in this Jira: FS-4768 (together with other changes for VS2012). > > /Peter > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Jo?o Mesquita > Skickat: den 9 januari 2013 14:07 > Till: FreeSWITCH Users Help > Kopia: freeswitch-users at lists.freeswitch.org > ?mne: Re: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project > > Dude, JIRA!! We will say that as many times as we need to aren't we? > > Sent from my iPhone > > On Jan 9, 2013, at 9:03 AM, Alexander Haugg wrote: > > Hi, > > the project libmpg123 do not build in debug mode. The error is in the ?Download mpg123.2012.vcxproj?. > see the fix in my git diff: > > diff --git libs/win32/Download mpg123.2012.vcxproj libs/win32/Download mpg123.2012.vcxproj > index ead3f9b..9942004 100644 > --- libs/win32/Download mpg123.2012.vcxproj > +++ libs/win32/Download mpg123.2012.vcxproj > @@ -68,7 +68,9 @@ > > Document > Downloading mpg123. > - if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." > + if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." > +if not exist "$(ProjectDir)..\libmpg123" move "$(ProjectDir)..\mpg123-1.14.4" "$(ProjectDir)..\libmpg123" > + > $(ProjectDir)..\libmpg123;%(Outputs) > Downloading mpg123. > if not exist "$(ProjectDir)..\libmpg123" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/mpg123-1.14.4.tar.bz2 "$(ProjectDir).." > > In the debug mode the mpg123-1.14.4 folder will not renamend to libmpg123. > > Is it possible to fix it? > > Thanks for your help. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > !DSPAM:50ed68b632765624898328! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/59ff2531/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image.jpeg Type: image/jpeg Size: 53248 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/59ff2531/attachment-0001.jpeg From krice at freeswitch.org Wed Jan 9 19:04:00 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 09 Jan 2013 10:04:00 -0600 Subject: [Freeswitch-users] SIP packet exceeds MTU? In-Reply-To: <50ED8E83.3030707@communicatefreely.net> Message-ID: Switch to TCP... This is why the RFCs tell you to support both UDP and TCP for SIP... On 1/9/13 9:36 AM, "Tim St. Pierre" wrote: > Hello, > > I'm having a problem with BLF in some cases, and I found that the length of > the username > was a factor in whether the lamps worked or not. > > When I did some packet captures, I found that problem packets were 1500 bytes > or so. > Since the endpoint is on a DSL line that has an MTU of 1460, the packet > doesn't fit. > > I'm already using compressed headers. Is there a way to reduce the packet > size more, or > is there another solution? I will include an example below: Note that my IP > address and > domain substutions may change the packet size a little, but the point is the > same. > > Thanks! > > NOTIFY sip:XXXX at YYY.YYY.YYY.YYY:5060;transport=udp SIP/2.0 > v:SIP/2.0/UDP zzz.zzz.zzz.zzz:5070;rport;branch=z9hG4bKD5QUr5m2Fm3Dp > Route:;transport=udp > Max-Forwards:70 > f:;tag=TrmHE4FBIrB4 > t:"Tim St. Pierre";tag=ecc2498bfd > i:3decd12decccafe1 > CSeq:448376002 NOTIFY > m: ip:yyy.yyy.yyy.yyy:61906%3Btransport%3Dudp> > User-Agent:Communicate Freely 2.0 > Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY, > PUBLISH,SUBSCRIBE > k:timer,precondition,path,replaces > o:dialog > u:talk,hold,conference,presence,dialog,line-seize,call-info,sla,include-sessio > n-description,presence.winfo,message-summary,refer > Subscription-State:active;expires=1137 > c:application/dialog-info+xml > l:639 > > > state="partial" > entity="sip:accountcode-701 at ourdomain.com"> > > confirmed > > display="park">sip:accountcode-701 at ourdomain.com;proto=park > > > > > > sip:accountcode-701 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From michel.brabants at gmail.com Wed Jan 9 19:07:16 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Wed, 9 Jan 2013 17:07:16 +0100 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: <1871B0BF-023F-4665-8398-0626556FA72F@jerris.com> References: <1871B0BF-023F-4665-8398-0626556FA72F@jerris.com> Message-ID: Hello, this is disabled in all profiles, but freeswitch keeps sending updates. I'm adding my profile-config here. I even added it to the gateway that is associated with the endpoint, which may be important .... The endpoint is a B2B sipserver itself, so I was wondering if this could be the problem ... I made some progress: when adding the following to the gateway-declaration, calls are disconnected after a minute (bit more), because after about 30 seconds freeswitch starts sending updates now ... This is really strange. I added the following to the gateway-settings to make calls atleast reach 1 minute: Why isn't freeswitch taking the param from the ua/profile that is associated with the profile? Thanks, Michel On Wed, Jan 9, 2013 at 2:21 PM, Michael Jerris wrote: > Its to update the caller id information. you can disable this: > > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#send-display-update > > Mike > > On Jan 9, 2013, at 4:48 AM, Michel Brabants > wrote: > > No, freeswitch is not getting a response as the client (B2B sipserver) > converts this to a reinvite with no sdp (which is incorrect of course). > However, if I need to wait in the client-developers to fix this, I'll be a > half year further... > > I was wondering however why freeswitch is sending these in the first place > as it contains no differences from previous sdp's. It isn't also sending > updates or reinvites to the other side. I don't see the purpose of these > updates. If it was to change the code, ... it would be better in this case > as freeswitch could send reinvites instead as these wouldn't be handled > incorrectly normally by the client. > > > Michel > > On Tue, Jan 8, 2013 at 10:11 PM, Steven Ayre wrote: > >> Yep my mistake... rather I meant is FS getting a response? >> (Wondering if FS is retransmitting and/or timing out the call on no >> response since you say it hangs up 30s later) >> >> >> >> On 8 January 2013 20:46, Kristian Kielhofner wrote: >> > The correct positive response to an UPDATE would be a 200 that is then >> > ACKd. See RFC 3311: >> > >> > http://tools.ietf.org/html/rfc3311 >> > >> > On Tue, Jan 8, 2013 at 2:40 PM, Steven Ayre >> wrote: >> >> Does the client send an ACK for the UPDATE? >> >> >> > >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/e64a32e0/attachment.html From msc at freeswitch.org Wed Jan 9 19:53:47 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Jan 2013 08:53:47 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Message-ID: Hello all! Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_09 We are going to be doing some community discussions on a few items such as the status of 1.2.6 and some ideas on how to assist with handling the load of Jira's. Talk to you shortly! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/5e63d0bc/attachment.html From fs-list at communicatefreely.net Wed Jan 9 19:53:56 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Wed, 09 Jan 2013 11:53:56 -0500 Subject: [Freeswitch-users] SIP packet exceeds MTU? In-Reply-To: References: Message-ID: <50EDA0A4.1010101@communicatefreely.net> Yes, I suppose I CAN switch to TCP, but I don't WANT to switch to TCP. My main gripe with TCP is that it holds a socket open for every registered endpoint. If I have to restart a profile, I have to leave it down for several minutes until all the sockets close before I can start it up again. I don't suppose there is a way to disable the and tags in the body? Thanks! Ken Rice wrote: > Switch to TCP... This is why the RFCs tell you to support both UDP and TCP > for SIP... > > > On 1/9/13 9:36 AM, "Tim St. Pierre" wrote: > >> Hello, >> >> I'm having a problem with BLF in some cases, and I found that the length of >> the username >> was a factor in whether the lamps worked or not. >> >> When I did some packet captures, I found that problem packets were 1500 bytes >> or so. >> Since the endpoint is on a DSL line that has an MTU of 1460, the packet >> doesn't fit. >> >> I'm already using compressed headers. Is there a way to reduce the packet >> size more, or >> is there another solution? I will include an example below: Note that my IP >> address and >> domain substutions may change the packet size a little, but the point is the >> same. >> >> Thanks! >> >> NOTIFY sip:XXXX at YYY.YYY.YYY.YYY:5060;transport=udp SIP/2.0 >> v:SIP/2.0/UDP zzz.zzz.zzz.zzz:5070;rport;branch=z9hG4bKD5QUr5m2Fm3Dp >> Route:;transport=udp >> Max-Forwards:70 >> f:;tag=TrmHE4FBIrB4 >> t:"Tim St. Pierre";tag=ecc2498bfd >> i:3decd12decccafe1 >> CSeq:448376002 NOTIFY >> m:> ip:yyy.yyy.yyy.yyy:61906%3Btransport%3Dudp> >> User-Agent:Communicate Freely 2.0 >> Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY, >> PUBLISH,SUBSCRIBE >> k:timer,precondition,path,replaces >> o:dialog >> u:talk,hold,conference,presence,dialog,line-seize,call-info,sla,include-sessio >> n-description,presence.winfo,message-summary,refer >> Subscription-State:active;expires=1137 >> c:application/dialog-info+xml >> l:639 >> >> >> > state="partial" >> entity="sip:accountcode-701 at ourdomain.com"> >> >> confirmed >> >> > display="park">sip:accountcode-701 at ourdomain.com;proto=park >> >> >> >> >> >> sip:accountcode-701 >> > >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > From jeff at jefflenk.com Wed Jan 9 20:46:28 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 9 Jan 2013 09:46:28 -0800 (PST) Subject: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project In-Reply-To: References: <1FFF97C269757C458224B7C895F35F151FBC15@cantor.std.visionutv.se> Message-ID: <1357753587971-7586164.post@n2.nabble.com> Thanks for checking the VS2012 build. Please update to git head and report the results to http://jira.freeswitch.org/browse/FS-4768 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Re-Bug-in-the-2012-solutin-in-the-libmpg123-project-tp7586153p7586164.html Sent from the freeswitch-users mailing list archive at Nabble.com. From drk at drkngs.net Wed Jan 9 21:10:09 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 09 Jan 2013 10:10:09 -0800 Subject: [Freeswitch-users] =?iso-8859-1?q?Bug_in_the_2012_solutin_in=09th?= =?iso-8859-1?q?e=09libmpg123=09project?= In-Reply-To: <1357753587971-7586164.post@n2.nabble.com> Message-ID: <20130109181009.2a05d522@mail.tritonwest.net> Jeff, you have to build the lib by hand since the projects that depend on it don't reference it. But if you build it by hand first it works. --Dave _____ From: Jeff Lenk [mailto:jeff at jefflenk.com] To: freeswitch-users at lists.freeswitch.org Sent: Wed, 09 Jan 2013 09:46:28 -0800 Subject: Re: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project Thanks for checking the VS2012 build. Please update to git head and report the results to http://jira.freeswitch.org/browse/FS-4768 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Re-Bug-in-the-2012-solutin-in-the-libmpg123-project-tp7586153p7586164.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/e465ca0b/attachment-0001.html From kris at kriskinc.com Wed Jan 9 22:13:02 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 9 Jan 2013 14:13:02 -0500 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: <2D95B81A-2356-43BE-9AC0-A2D0BDC741AB@edvina.net> References: <2D95B81A-2356-43BE-9AC0-A2D0BDC741AB@edvina.net> Message-ID: You are correct. I misread the call flow examples. The ACK was for the INVITE, not the update. With that being said the correct positive response to an UPDATE is still a 200, certainly not an ACK (even more so now). On Wed, Jan 9, 2013 at 3:29 AM, Olle E. Johansson wrote: > > 8 jan 2013 kl. 21:46 skrev Kristian Kielhofner : > >> The correct positive response to an UPDATE would be a 200 that is then >> ACKd. See RFC 3311: > The only transaction that needs an ACK is INVITE. > RFC3311 does not mention an ACK as a response to UPDATE. > > /O > >> >> http://tools.ietf.org/html/rfc3311 >> >> On Tue, Jan 8, 2013 at 2:40 PM, Steven Ayre wrote: >>> Does the client send an ACK for the UPDATE? >>> >> >> >> -- >> Kristian Kielhofner >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From mike at jerris.com Wed Jan 9 22:37:08 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 9 Jan 2013 14:37:08 -0500 Subject: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project In-Reply-To: <20130109181009.2a05d522@mail.tritonwest.net> References: <20130109181009.2a05d522@mail.tritonwest.net> Message-ID: <53952D88-AB5C-4FCB-8D02-7DE07630ACB7@jerris.com> Dave, if this is still the case with git head, please comment on the jira listed below. Mike On Jan 9, 2013, at 1:10 PM, Dave R. Kompel wrote: > Jeff, you have to build the lib by hand since the projects that depend on it don't reference it. But if you build it by hand first it works. > > --Dave > > From: Jeff Lenk [mailto:jeff at jefflenk.com] > To: freeswitch-users at lists.freeswitch.org > Sent: Wed, 09 Jan 2013 09:46:28 -0800 > Subject: Re: [Freeswitch-users] Bug in the 2012 solutin in the libmpg123 project > > Thanks for checking the VS2012 build. Please update to git head and report > the results to http://jira.freeswitch.org/browse/FS-4768 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/2dd60f88/attachment.html From sdevoy at bizfocused.com Wed Jan 9 22:48:24 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 9 Jan 2013 14:48:24 -0500 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! Message-ID: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> HI, I really hope this question is not as dumb as some of my recent questions (Anybody else wiped out you O/S lately?)! Anyway, I have an xml dial plan: I thought that would accept either 10 or 11 digit numbers starting with 1 and bridge to the gateway always using 1 plus the 10 digits. In fact in my log I can see: Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) [ten_or eleven_digiti_dial] destination_number(14104207777) =~ /^1?(\d{10})$/ break=on-false Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(ringback=${us-ring}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(ani=${outbound_caller_id_number}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(hangup_after_bridge=true) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action bridge(sofia/gateway/vitelity-inbound/14104207777) BUT It does not work for my customer : Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ /^1?(\d{10})$/ break=on-false Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(ringback=${us-ring}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(ani=${outbound_caller_id_number}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(hangup_after_bridge=true) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA LEADING 1 So, what am I missing? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/60c06530/attachment.html From jmesquita at freeswitch.org Wed Jan 9 22:59:57 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 9 Jan 2013 16:59:57 -0300 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> Message-ID: **** Jo?o Mesquit On Wed, Jan 9, 2013 at 5:48 PM, Sean Devoy wrote: > **** > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/8b3fdf8d/attachment-0001.html From djbinter at gmail.com Wed Jan 9 23:00:13 2013 From: djbinter at gmail.com (DJB International) Date: Wed, 9 Jan 2013 12:00:13 -0800 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> Message-ID: *1800336940 is 10-digits; therefore, it will match (\d{10}), which will be $1. $1 = 1800336940 * *Then, in your bridge you add 1 in front, so it is correct. * *What behaviors are you looking for in this case? * *-djbinter * On Wed, Jan 9, 2013 at 11:48 AM, Sean Devoy wrote: > HI,**** > > ** ** > > I really hope this question is not as dumb as some of my recent questions > (Anybody else wiped out you O/S lately?)!**** > > ** ** > > Anyway, I have an xml dial plan:**** > > **** > > **** > > **** > > **** > > * > *** > > **** > > **** > > **** > > **** > > **** > > ** ** > > I thought that would accept either 10 or 11 digit numbers starting with 1 > and bridge to the gateway always using 1 plus the 10 digits.**** > > ** ** > > In fact in my log I can see:**** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex *(PASS)*[ten_or eleven_digiti_dial] destination_number > *(14104207777*) =~ /^1?(\d{10})$/ break=on-false**** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ringback=${us-ring}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ani=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action bridge* > (sofia/gateway/vitelity-inbound/14104207777*)**** > > ** ** > > *BUT* > > It does not work for my customer :**** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex *(PASS*) > [ten_or_eleven_digiti_dial] destination_number(*1800336940*) =~ > /^1?(\d{10})$/ break=on-false**** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ringback=${us-ring}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ani=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action bridge* > (sofia/gateway/vitelity-inbound/11800336940*) NOTE THE EXTRA LEADING 1*** > * > > ** ** > > So, what am I missing?**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/e8342818/attachment.html From avi at avimarcus.net Wed Jan 9 23:02:28 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 9 Jan 2013 22:02:28 +0200 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> Message-ID: Ha, that was a tricky one! Here's the issue - he dialed: 1-800-336-940 -- that's too few digits! So your optional one made it match anyway. Your matching is too allowing - here's the regex I use: ^(?:1|\+1|001)?([2-9]\d\d[2-9]\d{6})$ That matches [nothing, 1, +1 or 001] prefix for USA. Then, bridge like now to "1$1". btw, if you want a separate match for USA toll-free it's: ^(?:1|\+1|001)? (8(00|55|66|77|88)[2-9]\d{6})$ There's also some good example on this page, the link goes directly to USA: http://wiki.freeswitch.org/wiki/Regular_Expression#NANPA_.2B1NxxNxxXXXX_E.164_Dialstring -Avi Marcus BestFone On Wed, Jan 9, 2013 at 9:48 PM, Sean Devoy wrote: > HI,**** > > ** ** > > I really hope this question is not as dumb as some of my recent questions > (Anybody else wiped out you O/S lately?)!**** > > ** ** > > Anyway, I have an xml dial plan:**** > > **** > > **** > > **** > > **** > > * > *** > > **** > > **** > > **** > > **** > > **** > > ** ** > > I thought that would accept either 10 or 11 digit numbers starting with 1 > and bridge to the gateway always using 1 plus the 10 digits.**** > > ** ** > > In fact in my log I can see:**** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex *(PASS)*[ten_or eleven_digiti_dial] destination_number > *(14104207777*) =~ /^1?(\d{10})$/ break=on-false**** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ringback=${us-ring}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ani=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action bridge* > (sofia/gateway/vitelity-inbound/14104207777*)**** > > ** ** > > *BUT* > > It does not work for my customer :**** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex *(PASS*) > [ten_or_eleven_digiti_dial] destination_number(*1800336940*) =~ > /^1?(\d{10})$/ break=on-false**** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ringback=${us-ring}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ani=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action bridge* > (sofia/gateway/vitelity-inbound/11800336940*) NOTE THE EXTRA LEADING 1*** > * > > ** ** > > So, what am I missing?**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/acee43d8/attachment-0001.html From jackal at cybershroud.net Wed Jan 9 23:02:53 2013 From: jackal at cybershroud.net (Carlos Flor) Date: Wed, 9 Jan 2013 15:02:53 -0500 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> Message-ID: Your customer dialed a 10 digit number, which was then converted to 11 by adding the 1. If you count the digits in the original dialed number, you will notice it's not a complete number. On Wed, Jan 9, 2013 at 2:48 PM, Sean Devoy wrote: > HI,**** > > ** ** > > I really hope this question is not as dumb as some of my recent questions > (Anybody else wiped out you O/S lately?)!**** > > ** ** > > Anyway, I have an xml dial plan:**** > > **** > > **** > > **** > > **** > > * > *** > > **** > > **** > > **** > > **** > > **** > > ** ** > > I thought that would accept either 10 or 11 digit numbers starting with 1 > and bridge to the gateway always using 1 plus the 10 digits.**** > > ** ** > > In fact in my log I can see:**** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex *(PASS)*[ten_or eleven_digiti_dial] destination_number > *(14104207777*) =~ /^1?(\d{10})$/ break=on-false**** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ringback=${us-ring}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ani=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action bridge* > (sofia/gateway/vitelity-inbound/14104207777*)**** > > ** ** > > *BUT* > > It does not work for my customer :**** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex *(PASS*) > [ten_or_eleven_digiti_dial] destination_number(*1800336940*) =~ > /^1?(\d{10})$/ break=on-false**** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ringback=${us-ring}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ani=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action bridge* > (sofia/gateway/vitelity-inbound/11800336940*) NOTE THE EXTRA LEADING 1*** > * > > ** ** > > So, what am I missing?**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/1f71023e/attachment.html From benoit.raymond at amiconcept.com Wed Jan 9 23:06:58 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Wed, 9 Jan 2013 15:06:58 -0500 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> Message-ID: <03c701cdeea4$f03959b0$d0ac0d10$@amiconcept.com> That?s because the 1800 you are showing has 10 digits, not 11. So the regex will keep the whole number even if it starts with a 1 because \d{10} condition is met. De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Sean Devoy Envoy? : 9 janvier 2013 14:48 ? : FreeSWITCH-users at lists.freeswitch.org Objet : [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! HI, I really hope this question is not as dumb as some of my recent questions (Anybody else wiped out you O/S lately?)! Anyway, I have an xml dial plan: I thought that would accept either 10 or 11 digit numbers starting with 1 and bridge to the gateway always using 1 plus the 10 digits. In fact in my log I can see: Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) [ten_or eleven_digiti_dial] destination_number(14104207777) =~ /^1?(\d{10})$/ break=on-false Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(ringback=${us-ring}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(ani=${outbound_caller_id_number}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(hangup_after_bridge=true) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action bridge(sofia/gateway/vitelity-inbound/14104207777) BUT It does not work for my customer : Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ /^1?(\d{10})$/ break=on-false Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(ringback=${us-ring}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(ani=${outbound_caller_id_number}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(hangup_after_bridge=true) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA LEADING 1 So, what am I missing? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/1d66e4cd/attachment-0001.html From victor.chukalovskiy at gmail.com Wed Jan 9 23:10:07 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Wed, 09 Jan 2013 15:10:07 -0500 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> Message-ID: <50EDCE9F.5070904@gmail.com> Sean, Your customer's destination number is one digit short. That's why it matches "1800336940" as \d{10} and prepends another "1" -Victor On 13-01-09 02:48 PM, Sean Devoy wrote: > > HI, > > I really hope this question is not as dumb as some of my recent > questions (Anybody else wiped out you O/S lately?)! > > Anyway, I have an xml dial plan: > > > > > > > > > > > > > > > > > > > > > > I thought that would accept either 10 or 11 digit numbers starting > with 1 and bridge to the gateway always using 1 plus the 10 digits. > > In fact in my log I can see: > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex *(PASS)* > [ten_or eleven_digiti_dial] destination_number*(14104207777*) =~ > /^1?(\d{10})$/ break=on-false > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ringback=${us-ring}) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ani=${outbound_caller_id_number}) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > bridge*(sofia/gateway/vitelity-inbound/14104207777*) > > *BUT* > > It does not work for my customer : > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex *(PASS*) > [ten_or_eleven_digiti_dial] destination_number(*1800336940*) =~ > /^1?(\d{10})$/ break=on-false > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ringback=${us-ring}) > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ani=${outbound_caller_id_number}) > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > bridge*(sofia/gateway/vitelity-inbound/11800336940*) NOTE THE EXTRA > LEADING 1 > > So, what am I missing? > > Thanks, > > Sean > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/db2f2754/attachment.html From steveayre at gmail.com Wed Jan 9 23:36:51 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 9 Jan 2013 20:36:51 +0000 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> Message-ID: They're dialing 10 digits including the 1. '1' plus 10 digits can't match, it's not long enough, So the regex instead ignores the optional '1' and tries to match just \d{10}, which does match since that allows the first digit to be a 1. If you want to handle '1' plus 9 digits differently, then your regex will need tweaking. On 9 January 2013 19:48, Sean Devoy wrote: > HI, > > > > I really hope this question is not as dumb as some of my recent questions > (Anybody else wiped out you O/S lately?)! > > > > Anyway, I have an xml dial plan: > > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > > > > data="effective_caller_id_name=${outbound_caller_id_name}"/> > > > > data="sofia/gateway/vitelity-inbound/1$1"/> > > > > > > > > I thought that would accept either 10 or 11 digit numbers starting with 1 > and bridge to the gateway always using 1 plus the 10 digits. > > > > In fact in my log I can see: > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) [ten_or > eleven_digiti_dial] destination_number(14104207777) =~ /^1?(\d{10})$/ > break=on-false > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ringback=${us-ring}) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ani=${outbound_caller_id_number}) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > bridge(sofia/gateway/vitelity-inbound/14104207777) > > > > BUT > > It does not work for my customer : > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) > [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ /^1?(\d{10})$/ > break=on-false > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ringback=${us-ring}) > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ani=${outbound_caller_id_number}) > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA LEADING 1 > > > > So, what am I missing? > > > > Thanks, > > Sean > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From daniel at pocock.com.au Wed Jan 9 23:44:36 2013 From: daniel at pocock.com.au (Daniel Pocock) Date: Wed, 09 Jan 2013 21:44:36 +0100 Subject: [Freeswitch-users] FOSDEM 2013 - Feb 2-3 - VoIP in the main track Message-ID: <50EDD6B4.5070800@pocock.com.au> Hi all, VoIP and RTC is in the main track at FOSDEM this year. I've been involved in preparing a presentation that is not focusing on any specific product, rather, we are looking at the bigger picture and how to get open source products working together on a scale that will overcome those proprietary solutions that many people find unpleasant: https://fosdem.org/2013/schedule/event/free_open_secure_communications/ There is also an interview giving some more background: https://fosdem.org/2013/interviews/2013-daniel-pocock-and-peter-saint-andre-and-simon-tennant-and-evan-prodromou-and-daniel-constantin-mierla-and-emil-ivov/ There is also an XMPP devroom on Saturday and a telephony devroom on Sunday - schedules will be published shortly. I hope to meet some of you at FOSDEM Regards, Daniel From bdfoster at endigotech.com Wed Jan 9 23:53:36 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 9 Jan 2013 15:53:36 -0500 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> Message-ID: We have two different regex's/dialplans to handle this issue. One that matches 10 digits and one that matches 11. I think freeswitch example configs cover this. Sent from my iPhone On Jan 9, 2013, at 3:36 PM, Steven Ayre wrote: > They're dialing 10 digits including the 1. > > '1' plus 10 digits can't match, it's not long enough, > > So the regex instead ignores the optional '1' and tries to match just > \d{10}, which does match since that allows the first digit to be a 1. > > If you want to handle '1' plus 9 digits differently, then your regex > will need tweaking. > > > > > > On 9 January 2013 19:48, Sean Devoy wrote: >> HI, >> >> >> >> I really hope this question is not as dumb as some of my recent questions >> (Anybody else wiped out you O/S lately?)! >> >> >> >> Anyway, I have an xml dial plan: >> >> >> >> >> >> >> >> > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> >> >> >> > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> >> >> >> > data="sofia/gateway/vitelity-inbound/1$1"/> >> >> >> >> >> >> >> >> I thought that would accept either 10 or 11 digit numbers starting with 1 >> and bridge to the gateway always using 1 plus the 10 digits. >> >> >> >> In fact in my log I can see: >> >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) [ten_or >> eleven_digiti_dial] destination_number(14104207777) =~ /^1?(\d{10})$/ >> break=on-false >> >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(ringback=${us-ring}) >> >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(effective_caller_id_number=${outbound_caller_id_number}) >> >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(ani=${outbound_caller_id_number}) >> >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(effective_caller_id_name=${outbound_caller_id_name}) >> >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(hangup_after_bridge=true) >> >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> bridge(sofia/gateway/vitelity-inbound/14104207777) >> >> >> >> BUT >> >> It does not work for my customer : >> >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) >> [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ /^1?(\d{10})$/ >> break=on-false >> >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >> set(ringback=${us-ring}) >> >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >> set(effective_caller_id_number=${outbound_caller_id_number}) >> >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >> set(ani=${outbound_caller_id_number}) >> >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >> set(effective_caller_id_name=${outbound_caller_id_name}) >> >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(hangup_after_bridge=true) >> >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >> bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA LEADING 1 >> >> >> >> So, what am I missing? >> >> >> >> Thanks, >> >> Sean >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From avi at avimarcus.net Thu Jan 10 00:00:59 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 9 Jan 2013 23:00:59 +0200 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> Message-ID: @Brian that unnecessarily complicates things, if you check my previous email you can easily do it in one extension. -Avi Marcus BestFone On Wed, Jan 9, 2013 at 10:53 PM, Brian Foster wrote: > We have two different regex's/dialplans to handle this issue. One that > matches 10 digits and one that matches 11. I think freeswitch example > configs cover this. > > Sent from my iPhone > > On Jan 9, 2013, at 3:36 PM, Steven Ayre wrote: > > > They're dialing 10 digits including the 1. > > > > '1' plus 10 digits can't match, it's not long enough, > > > > So the regex instead ignores the optional '1' and tries to match just > > \d{10}, which does match since that allows the first digit to be a 1. > > > > If you want to handle '1' plus 9 digits differently, then your regex > > will need tweaking. > > > > > > > > > > > > On 9 January 2013 19:48, Sean Devoy wrote: > >> HI, > >> > >> > >> > >> I really hope this question is not as dumb as some of my recent > questions > >> (Anybody else wiped out you O/S lately?)! > >> > >> > >> > >> Anyway, I have an xml dial plan: > >> > >> > >> > >> > >> > >> > >> > >> >> data="effective_caller_id_number=${outbound_caller_id_number}"/> > >> > >> > >> > >> >> data="effective_caller_id_name=${outbound_caller_id_name}"/> > >> > >> > >> > >> >> data="sofia/gateway/vitelity-inbound/1$1"/> > >> > >> > >> > >> > >> > >> > >> > >> I thought that would accept either 10 or 11 digit numbers starting with > 1 > >> and bridge to the gateway always using 1 plus the 10 digits. > >> > >> > >> > >> In fact in my log I can see: > >> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) > [ten_or > >> eleven_digiti_dial] destination_number(14104207777) =~ /^1?(\d{10})$/ > >> break=on-false > >> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(ringback=${us-ring}) > >> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(effective_caller_id_number=${outbound_caller_id_number}) > >> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(ani=${outbound_caller_id_number}) > >> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(effective_caller_id_name=${outbound_caller_id_name}) > >> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(hangup_after_bridge=true) > >> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> bridge(sofia/gateway/vitelity-inbound/14104207777) > >> > >> > >> > >> BUT > >> > >> It does not work for my customer : > >> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) > >> [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ > /^1?(\d{10})$/ > >> break=on-false > >> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >> set(ringback=${us-ring}) > >> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >> set(effective_caller_id_number=${outbound_caller_id_number}) > >> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >> set(ani=${outbound_caller_id_number}) > >> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >> set(effective_caller_id_name=${outbound_caller_id_name}) > >> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(hangup_after_bridge=true) > >> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >> bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA > LEADING 1 > >> > >> > >> > >> So, what am I missing? > >> > >> > >> > >> Thanks, > >> > >> Sean > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/5c6caf25/attachment.html From lconroy at insensate.co.uk Thu Jan 10 00:27:57 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Wed, 9 Jan 2013 21:27:57 +0000 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: <2ef401cdee82$090f9df0$1b2ed9d0$@bizfocused.com> References: <50DE75C4.1030500@communicatefreely.net> <1ADA846F-0D19-4FE3-AFA3-7CA5E8E4C127@insensate.co.uk> <2ef401cdee82$090f9df0$1b2ed9d0$@bizfocused.com> Message-ID: Hi Sean, folks, OK -- this is drifting away from the purely technical, but ... It depends on whether or not you trust your customers to have clue, and on whether or not your customers have sensible DNS providers. IMHO, most DNS providers are the sticking point -- they wouldn't know a NAPTR if it jumped up and bit them on the t*t, and have less desire to change their customer dashboard than pretty much anything; if it works for customer web sites, why would they need to do anything else? In theory, your customers could have their own domain, put a single NAPTR in that (pointing at a SRV you set up in your DNS domain), and you could then put your SIP servers at any FQDN you'd like, pointed to by those SRVs. Remember, the customer's domain can have a NAPTR but no A record and no SRV -- the NAPTR can quite easily point to a SRV in an entirely different domain, just as the SRV can point to a FQDN in yet another domain. There's a lot of flexibility there. In this ideal world, the customer gets to use , you keep your SRVs under your control in your domain and can tweak these at need without any change needed to customer domain's DNS records. Thus customers don't even need to know in advance what the IP address(es) of your servers are. It's all good. You (obviously) do need to know what sipdomains your customers are going to use and to configure fS accordingly to serve/auth these, but that's about it. BUT ... 1. Some old SIP implementations are crock, and don't understand NAPTRs either. These cobbled together apologies for code just AssUMe that there's going to be a SRV where they expect it. Some new SIP implementations are equally "challenged". 2. You would be wise to check that each of your customers DID actually manage to put a NAPTR into their domain (i.e., they knew exactly what to put into DNS because you told them, and their DNS provider/IT department had the clue and the mechanism to make this so). It's pretty easy to check from anywhere (dig is your friend :), but it helps your customer care staff to know (when a customer calls up with a problem) that the customer may well have shot themselves in the foot. So ... in practice, it's useful for them to put a SRV in their domain as well. Doesn't solve problem 2, but will avoid most of problem 1. As for giving them their own FQDN -- (in your domain): yup, you can do that as well. It does mean that you need to be busy when you need to swap machines out or change machine addresses because some rag'eads are DDoSing you (i.e. if it all goes pete tong every one of your customer FQDNs' A/AAA records all need to be updated ASAP). Personally I'd still have sub-domains with NAPTRs and SRVs pointing to other sub-domains of yours where your server hostnames and their A/AAA records live. It really depends on how much your customers want to have their own brand/identity, and how much you can trust them (and their service providers) to do clueful things. all the best, Lawrence On 9 Jan 2013, at 15:57, Sean Devoy wrote: > Great info Lawrence, especially the MX analogy. Thanks. > > I have to take this one more step and ask a question. We have FS configured > for multitenant. Of course the "domains" for the tenants are FQDNs. I > found it simple to setup and wonderful for setting up customers as their own > virtual switches. > > Clearly this is a case where the sip address has to contain the FQDN. (It > also gives me a chuckle every day when I see those a-holes' registration > failures for "100@" when a name is required!) > > My FS Domains are like "fs_A.xyz.com" and "FS_B.xyz.com" which I currently > have as A records with the same IP address. > If I add an SRV record for _sip._udp.xyz.com to that IP Address, are you > saying I won't need the A records? Or do I need SRV records for each > sub-domain? > > Of course this is assuming I set my hardware (Cisco SPA 5xx) to "Use SRV? > Yes" > > Thanks for the information. > > Sean > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lawrence > Conroy > Sent: Tuesday, January 08, 2013 7:05 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Best practices question about SIP > registration > > Hi again Steven, > with apologies for top-posting ... > Re. flowroute.com > Many of us have sup-domains for different classes of users -- looks like > your sub-domain is sip.flowroute.com. > > However, flowroute.com does have a NAPTR (it has two): > ;; ANSWER SECTION: > flowroute.com. 21600 IN NAPTR 100 10 "s" "SIP+D2U" "" > _sip._udp.flowroute.com. > flowroute.com. 21600 IN NAPTR 102 20 "s" "SIP+D2T" "" > _sip._tcp.flowroute.com. > > Also, the _sip._udp.flowroute SRV shows: > ;; ANSWER SECTION: > _sip._udp.flowroute.com. 43200 IN SRV 20 10 5060 > sip-ca1.flowroute.com. > _sip._udp.flowroute.com. 43200 IN SRV 10 10 5060 > sip-nv1.flowroute.com. > > as does the _sip.tcp.flowroute.com SRV: > ;; ANSWER SECTION: > _sip._tcp.flowroute.com. 43200 IN SRV 10 10 5060 > sip-nv1.flowroute.com. > _sip._tcp.flowroute.com. 43200 IN SRV 20 10 5060 > sip-ca1.flowroute.com. > > so -- if a client is trying to call to sip:1000 at flowroute.com, that client > looks up the NAPTR set for flowroute.com (the sip domainpart), gets back the > two entries, and chooses to look for a SRV under _sip._udp.flowroute.com (if > it wants to call using UDP) or _sip.tcp.flowroute.com (if it wants to call > using TCP or use sips:). > flowroute would prefer that you contacted them by UDP (it has a lower > preference -- 10 versus the D2T's 20). > If you do contact them, they'd prefer that you used sip-nv1.flowroute.com. > (as that has a better/lower preference than sip-cal). > > On to the polycoms -- I don't have any of these, so I'm guessing, but ... > It looks strongly like the registration server is at fs.domain.local, which > has an IP address of 10.10.10.11 for the DNS-challenged, and is listening on > port 5060. > The AuthID/UserID for this 'phone to use 1000, which is kinda apparent from > the SIP address the phone has, which is sip:1000 at fs.domain.local. > I'd AssUMe that there's also a password field :). > > So ... if you had mydomain.com, and for your internal use you had a > sub-domain of, say, internal.mydomain.com, and you had one fS handling your > local 'phones, running on a machine called fs.internal.domain.com which had > an IP address of 10.10.10.11, I'd put into the local view of DNS: > > internal.mydomain.com. IN NAPTR 100 10 "s" "SIP+D2U" "" > _sip._udp.internal.mydomain.com. > _sip._udp.internal.mydomain.com. IN SRV 10 10 5060 fs.internal.mydomain.com > fs.internal.mydomain.com. IN A 10.10.10.11 > > Note that I'm assuming that you'd be using a split-view DNS (i.e., your > local DNS server would only give these answers to its local machines) as > it's kinda impolite to give private addresses like 10.10.10.11 to the > outside world -- clients are sometimes too dumb to realise they can't > contact that server using this address. > [...which is why we have one way audio -- stupid implementers :]p > > Hope that helps, > Lawrence > > On 8 Jan 2013, at 23:02, Steven Schoch wrote: >> On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy > wrote: >> >>> Hi there, >>> at the risk of butting in on someone else's party ... >>> Nope; your interpretations is NOT best practice. >>> >> >> This party is open to all. I appreciate these answers, and as usual I >> have noticed that the fastest way to learn is to do it wrong, and then >> have somebody correct me! Thanks! >> >> >>> With SIP (see RFC 3263), you do a lookup on the SRV record (at >>> _sip._udp.) to find the machine that handles SIP >>> registrations/incalls for the domain. That also gives you the port on >>> which that machine is listening. >>> >> >> For example, I did a lookup: >> >> $ dig SRV _sip._udp.sip.flowroute.com. >> >> This returns: >> >> _sip._udp.sip.flowroute.com. 43200 IN SRV 20 10 5060 >> sip-ca1.flowroute.com. >> _sip._udp.sip.flowroute.com. 43200 IN SRV 10 10 5060 >> sip-nv1.flowroute.com. >> >> >> >>> (Yup, you can also have a NAPTR record in the domain to tell you >>> where the SRV record is, but many folks don't bother -- for Best >>> Practice, you should, but ...) >>> >> >> $ dig NAPTR sip.flowroute.com >> >> sip.flowroute.com. 42818 IN NAPTR 100 10 "s" "SIP+D2U" "" >> _sip._udp.sip.flowroute.com. >> sip.flowroute.com. 42818 IN NAPTR 102 20 "s" "SIP+D2U" "" >> _sip._tcp.sip.flowroute.com. >> >> I found that _sip._tcp.sip.flowroute.com does not have a SRV record. >> I don't know what that means. The RFC says that you should use >> "SIP+D2T" for TCP. >> >> Now, as this relates to the Polycom phone example, there are these fields: >> >> reg.1.address="1000 at fs.domain.local" >> reg.1.auth.userId="1000" >> reg.1.server.1.address="10.10.10.11" >> reg.1.server.1.port="5060" >> >> Since this is a VoIP phone that loads its configuration from the >> server, not VoIP software that is configured by the user, it really >> only needs the IP address, so the server address field is kind of moot. >> >> But the example domain here is "fs.domain.local". Why not just >> "domain.local"? Why didn't Flowroute use "flowroute.com" instead of " >> sip.flowroute.com"? >> >> And my other question was how is the address "1000 at fs.domain.local" used? >> Does FreeSwitch use that or does it just use the auth.userId? >> >> -- >> Steve >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Thu Jan 10 00:34:02 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Jan 2013 13:34:02 -0800 Subject: [Freeswitch-users] New FreeSWITCH sound prompts Message-ID: Hello all! I'm preparing a new order for English sound prompts. If you any thoughts about what is missing please let me know. For those who may be new to the project, the sound files are all documented in $fs_src/docs/phrase/phrase_en.xml. There are other documents in there as well for the other languages. Feel free to translate our prompts into your favorite tongue! The order will go in Monday the 14th by end of the day. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/392c58ca/attachment.html From krice at freeswitch.org Thu Jan 10 00:33:57 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 09 Jan 2013 15:33:57 -0600 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: Message-ID: If you are trying to filter NANPA numbers why not use a tighter REGEX like ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6})$ This would leave the base 10 digit number in $1 On 1/9/13 2:00 PM, "DJB International" wrote: > 1800336940 is 10-digits; therefore, it will match (\d{10}), which will be $1.? > > $1 = 1800336940 > > Then, in your bridge you add 1 in front, so it is correct. > > What behaviors are you looking for in this case? > > -djbinter > > > On Wed, Jan 9, 2013 at 11:48 AM, Sean Devoy wrote: >> HI, >> ? >> I really hope this question is not as dumb as some of my recent questions >> (Anybody else wiped out you O/S lately?)! >> ? >> Anyway,? I have an xml dial plan: >> ? >> ??? >> ????? >> ????? > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> ????? >> ????? > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> ????? >> ????? > data="sofia/gateway/vitelity-inbound/1$1"/> >> ??? >> ? >> ? >> I thought that would accept either 10 or 11 digit numbers starting with 1 and >> bridge to the gateway always using 1 plus the 10 digits. >> ? >> In fact in my log I can see: >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) [ten_or >> eleven_digiti_dial] destination_number(14104207777 ) =~ >> /^1?(\d{10})$/ break=on-false >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(ringback=${us-ring}) >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(effective_caller_id_number=${outbound_caller_id_number}) >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(ani=${outbound_caller_id_number}) >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(effective_caller_id_name=${outbound_caller_id_name}) >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(hangup_after_bridge=true) >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> bridge(sofia/gateway/vitelity-inbound/14104207777 ) >> ? >> BUT >> It does not work for my customer : >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) >> [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ /^1?(\d{10})$/ >> break=on-false >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >> set(ringback=${us-ring}) >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >> set(effective_caller_id_number=${outbound_caller_id_number}) >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >> set(ani=${outbound_caller_id_number}) >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >> set(effective_caller_id_name=${outbound_caller_id_name}) >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >> set(hangup_after_bridge=true) >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >> bridge(sofia/gateway/vitelity-inbound/11800336940)? NOTE THE EXTRA LEADING 1 >> ? >> So, what am I missing? >> ? >> Thanks, >> Sean >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/7a6d055d/attachment.html From sos at sokhapkin.dyndns.org Thu Jan 10 00:44:25 2013 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 09 Jan 2013 16:44:25 -0500 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: References: Message-ID: <2905110.foZzicuv5A@sos> It's better to remove trailing "$", there could be more digits on handset's redial after entering DTMF codes on previous call. On Wednesday 09 January 2013 15:33:57 Ken Rice wrote: > If you are trying to filter NANPA numbers why not use a tighter REGEX like > ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6})$ > > This would leave the base 10 digit number in $1 > > On 1/9/13 2:00 PM, "DJB International" wrote: > > 1800336940 is 10-digits; therefore, it will match (\d{10}), which will be > > $1. > > > > $1 = 1800336940 > > > > Then, in your bridge you add 1 in front, so it is correct. > > > > What behaviors are you looking for in this case? > > > > -djbinter > > > > On Wed, Jan 9, 2013 at 11:48 AM, Sean Devoy wrote: > >> HI, > >> > >> I really hope this question is not as dumb as some of my recent questions > >> (Anybody else wiped out you O/S lately?)! > >> > >> Anyway, I have an xml dial plan: > >> > >> > >> > >> >> data="effective_caller_id_number=${outbound_caller_id_number}"/> > >> > >> >> data="effective_caller_id_name=${outbound_caller_id_name}"/> > >> > >> >> data="sofia/gateway/vitelity-inbound/1$1"/> > >> > >> > >> > >> I thought that would accept either 10 or 11 digit numbers starting with 1 > >> and bridge to the gateway always using 1 plus the 10 digits. > >> > >> In fact in my log I can see: > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) [ten_or > >> eleven_digiti_dial] destination_number(14104207777 ) > >> =~ /^1?(\d{10})$/ break=on-false > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(ringback=${us-ring}) > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(effective_caller_id_number=${outbound_caller_id_number}) > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(ani=${outbound_caller_id_number}) > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(effective_caller_id_name=${outbound_caller_id_name}) > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(hangup_after_bridge=true) > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> bridge(sofia/gateway/vitelity-inbound/14104207777 ) > >> > >> BUT > >> It does not work for my customer : > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) > >> [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ > >> /^1?(\d{10})$/ break=on-false > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >> set(ringback=${us-ring}) > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >> set(effective_caller_id_number=${outbound_caller_id_number}) > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >> set(ani=${outbound_caller_id_number}) > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >> set(effective_caller_id_name=${outbound_caller_id_name}) > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >> set(hangup_after_bridge=true) > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >> bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA > >> LEADING 1 > >> So, what am I missing? > >> > >> Thanks, > >> Sean > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org From krice at freeswitch.org Thu Jan 10 01:00:00 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 09 Jan 2013 16:00:00 -0600 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: <2905110.foZzicuv5A@sos> Message-ID: We can agree to disagree on that... I would argue if that's happening they need to update their handset... Its not 1985 any more On 1/9/13 3:44 PM, "Sergey Okhapkin" wrote: > It's better to remove trailing "$", there could be more digits on handset's > redial after entering DTMF codes on previous call. > > On Wednesday 09 January 2013 15:33:57 Ken Rice wrote: >> If you are trying to filter NANPA numbers why not use a tighter REGEX like >> ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6})$ >> >> This would leave the base 10 digit number in $1 >> >> On 1/9/13 2:00 PM, "DJB International" wrote: >>> 1800336940 is 10-digits; therefore, it will match (\d{10}), which will be >>> $1. >>> >>> $1 = 1800336940 >>> >>> Then, in your bridge you add 1 in front, so it is correct. >>> >>> What behaviors are you looking for in this case? >>> >>> -djbinter >>> >>> On Wed, Jan 9, 2013 at 11:48 AM, Sean Devoy wrote: >>>> HI, >>>> >>>> I really hope this question is not as dumb as some of my recent questions >>>> (Anybody else wiped out you O/S lately?)! >>>> >>>> Anyway, I have an xml dial plan: >>>> >>>> >>>> >>>> >>> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>>> >>>> >>> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>>> >>>> >>> data="sofia/gateway/vitelity-inbound/1$1"/> >>>> >>>> >>>> >>>> I thought that would accept either 10 or 11 digit numbers starting with 1 >>>> and bridge to the gateway always using 1 plus the 10 digits. >>>> >>>> In fact in my log I can see: >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) [ten_or >>>> eleven_digiti_dial] destination_number(14104207777 ) >>>> =~ /^1?(\d{10})$/ break=on-false >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>> set(ringback=${us-ring}) >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>> set(effective_caller_id_number=${outbound_caller_id_number}) >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>> set(ani=${outbound_caller_id_number}) >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>> set(effective_caller_id_name=${outbound_caller_id_name}) >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>> set(hangup_after_bridge=true) >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>> bridge(sofia/gateway/vitelity-inbound/14104207777 ) >>>> >>>> BUT >>>> It does not work for my customer : >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) >>>> [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ >>>> /^1?(\d{10})$/ break=on-false >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >>>> set(ringback=${us-ring}) >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >>>> set(effective_caller_id_number=${outbound_caller_id_number}) >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >>>> set(ani=${outbound_caller_id_number}) >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >>>> set(effective_caller_id_name=${outbound_caller_id_name}) >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>> set(hangup_after_bridge=true) >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >>>> bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA >>>> LEADING 1 >>>> So, what am I missing? >>>> >>>> Thanks, >>>> Sean >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From djbinter at gmail.com Thu Jan 10 01:03:07 2013 From: djbinter at gmail.com (DJB International) Date: Wed, 9 Jan 2013 14:03:07 -0800 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: <2905110.foZzicuv5A@sos> References: <2905110.foZzicuv5A@sos> Message-ID: Ken, Is it correct that in the group but not capture (?: we will need an escape character \ to match the literal "+" character as well. So, I think it should be: ^(?:\+1|1)?([2-9]\d{2}[2-9]\d{6})$ -djbinter On Wed, Jan 9, 2013 at 1:44 PM, Sergey Okhapkin wrote: > It's better to remove trailing "$", there could be more digits on handset's > redial after entering DTMF codes on previous call. > > On Wednesday 09 January 2013 15:33:57 Ken Rice wrote: > > If you are trying to filter NANPA numbers why not use a tighter REGEX > like > > ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6})$ > > > > This would leave the base 10 digit number in $1 > > > > On 1/9/13 2:00 PM, "DJB International" wrote: > > > 1800336940 is 10-digits; therefore, it will match (\d{10}), which will > be > > > $1. > > > > > > $1 = 1800336940 > > > > > > Then, in your bridge you add 1 in front, so it is correct. > > > > > > What behaviors are you looking for in this case? > > > > > > -djbinter > > > > > > On Wed, Jan 9, 2013 at 11:48 AM, Sean Devoy > wrote: > > >> HI, > > >> > > >> I really hope this question is not as dumb as some of my recent > questions > > >> (Anybody else wiped out you O/S lately?)! > > >> > > >> Anyway, I have an xml dial plan: > > >> > > >> > > >> > > >> > >> data="effective_caller_id_number=${outbound_caller_id_number}"/> > > >> data="ani=${outbound_caller_id_number}"/> > > >> > >> data="effective_caller_id_name=${outbound_caller_id_name}"/> > > >> > > >> > >> data="sofia/gateway/vitelity-inbound/1$1"/> > > >> > > >> > > >> > > >> I thought that would accept either 10 or 11 digit numbers starting > with 1 > > >> and bridge to the gateway always using 1 plus the 10 digits. > > >> > > >> In fact in my log I can see: > > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) > [ten_or > > >> eleven_digiti_dial] destination_number(14104207777 ) > > >> =~ /^1?(\d{10})$/ break=on-false > > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > > >> set(ringback=${us-ring}) > > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > > >> set(effective_caller_id_number=${outbound_caller_id_number}) > > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > > >> set(ani=${outbound_caller_id_number}) > > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > > >> set(effective_caller_id_name=${outbound_caller_id_name}) > > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > > >> set(hangup_after_bridge=true) > > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > > >> bridge(sofia/gateway/vitelity-inbound/14104207777 ) > > >> > > >> BUT > > >> It does not work for my customer : > > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) > > >> [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ > > >> /^1?(\d{10})$/ break=on-false > > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > > >> set(ringback=${us-ring}) > > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > > >> set(effective_caller_id_number=${outbound_caller_id_number}) > > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > > >> set(ani=${outbound_caller_id_number}) > > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > > >> set(effective_caller_id_name=${outbound_caller_id_name}) > > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > > >> set(hangup_after_bridge=true) > > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > > >> bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA > > >> LEADING 1 > > >> So, what am I missing? > > >> > > >> Thanks, > > >> Sean > > >> > > >> > _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/54176ec8/attachment.html From sdevoy at bizfocused.com Thu Jan 10 01:07:24 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 9 Jan 2013 17:07:24 -0500 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: References: Message-ID: <336b01cdeeb5$b47992d0$1d6cb870$@bizfocused.com> THANK YOU TO ALL who pointed out the primary problem - the user dialed too few actual digits, Oh to be young and be able to see well again. Thanks to Avi and Ken for the improved US N.A. regex. CRAP - I went to add it to the WIKI and it's in there on the Regular Expression Page. Thanks everyone. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Wednesday, January 09, 2013 4:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! If you are trying to filter NANPA numbers why not use a tighter REGEX like ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6})$ This would leave the base 10 digit number in $1 On 1/9/13 2:00 PM, "DJB International" wrote: 1800336940 is 10-digits; therefore, it will match (\d{10}), which will be $1. $1 = 1800336940 Then, in your bridge you add 1 in front, so it is correct. What behaviors are you looking for in this case? -djbinter On Wed, Jan 9, 2013 at 11:48 AM, Sean Devoy wrote: HI, I really hope this question is not as dumb as some of my recent questions (Anybody else wiped out you O/S lately?)! Anyway, I have an xml dial plan: I thought that would accept either 10 or 11 digit numbers starting with 1 and bridge to the gateway always using 1 plus the 10 digits. In fact in my log I can see: Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) [ten_or eleven_digiti_dial] destination_number(14104207777 ) =~ /^1?(\d{10})$/ break=on-false Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(ringback=${us-ring}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(ani=${outbound_caller_id_number}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(hangup_after_bridge=true) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action bridge(sofia/gateway/vitelity-inbound/14104207777 ) BUT It does not work for my customer : Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ /^1?(\d{10})$/ break=on-false Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(ringback=${us-ring}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(ani=${outbound_caller_id_number}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(hangup_after_bridge=true) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA LEADING 1 So, what am I missing? Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/6957e0fc/attachment-0001.html From msc at freeswitch.org Thu Jan 10 01:07:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Jan 2013 14:07:30 -0800 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: References: <2905110.foZzicuv5A@sos> Message-ID: Or you could both win by modifying the regex a bit: ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6})\d*$ That assumes a valid NANPA number with "some extra digits" is just that - a valid NANPA number with some extra digits. However, I wouldn't trust a handset that doesn't know the difference between digits in the phone number and digits sent after far end answered. -MC On Wed, Jan 9, 2013 at 2:00 PM, Ken Rice wrote: > We can agree to disagree on that... I would argue if that's happening they > need to update their handset... Its not 1985 any more > > > On 1/9/13 3:44 PM, "Sergey Okhapkin" wrote: > > > It's better to remove trailing "$", there could be more digits on > handset's > > redial after entering DTMF codes on previous call. > > > > On Wednesday 09 January 2013 15:33:57 Ken Rice wrote: > >> If you are trying to filter NANPA numbers why not use a tighter REGEX > like > >> ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6})$ > >> > >> This would leave the base 10 digit number in $1 > >> > >> On 1/9/13 2:00 PM, "DJB International" wrote: > >>> 1800336940 is 10-digits; therefore, it will match (\d{10}), which will > be > >>> $1. > >>> > >>> $1 = 1800336940 > >>> > >>> Then, in your bridge you add 1 in front, so it is correct. > >>> > >>> What behaviors are you looking for in this case? > >>> > >>> -djbinter > >>> > >>> On Wed, Jan 9, 2013 at 11:48 AM, Sean Devoy > wrote: > >>>> HI, > >>>> > >>>> I really hope this question is not as dumb as some of my recent > questions > >>>> (Anybody else wiped out you O/S lately?)! > >>>> > >>>> Anyway, I have an xml dial plan: > >>>> > >>>> > >>>> > >>>> >>>> data="effective_caller_id_number=${outbound_caller_id_number}"/> > >>>> data="ani=${outbound_caller_id_number}"/> > >>>> >>>> data="effective_caller_id_name=${outbound_caller_id_name}"/> > >>>> > >>>> >>>> data="sofia/gateway/vitelity-inbound/1$1"/> > >>>> > >>>> > >>>> > >>>> I thought that would accept either 10 or 11 digit numbers starting > with 1 > >>>> and bridge to the gateway always using 1 plus the 10 digits. > >>>> > >>>> In fact in my log I can see: > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) > [ten_or > >>>> eleven_digiti_dial] destination_number(14104207777 > ) > >>>> =~ /^1?(\d{10})$/ break=on-false > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(ringback=${us-ring}) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(effective_caller_id_number=${outbound_caller_id_number}) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(ani=${outbound_caller_id_number}) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(effective_caller_id_name=${outbound_caller_id_name}) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(hangup_after_bridge=true) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> bridge(sofia/gateway/vitelity-inbound/14104207777 ) > >>>> > >>>> BUT > >>>> It does not work for my customer : > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) > >>>> [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ > >>>> /^1?(\d{10})$/ break=on-false > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >>>> set(ringback=${us-ring}) > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >>>> set(effective_caller_id_number=${outbound_caller_id_number}) > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >>>> set(ani=${outbound_caller_id_number}) > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >>>> set(effective_caller_id_name=${outbound_caller_id_name}) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(hangup_after_bridge=true) > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >>>> bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA > >>>> LEADING 1 > >>>> So, what am I missing? > >>>> > >>>> Thanks, > >>>> Sean > >>>> > >>>> > _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/39de7981/attachment.html From sos at sokhapkin.dyndns.org Thu Jan 10 01:07:52 2013 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Wed, 09 Jan 2013 17:07:52 -0500 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: References: Message-ID: <2371457.fKI3iQqlDd@sos> My Uniden handset made around 2010. It sends extra DTMF digits on redial. On Wednesday 09 January 2013 16:00:00 Ken Rice wrote: > We can agree to disagree on that... I would argue if that's happening they > need to update their handset... Its not 1985 any more > > On 1/9/13 3:44 PM, "Sergey Okhapkin" wrote: > > It's better to remove trailing "$", there could be more digits on > > handset's > > redial after entering DTMF codes on previous call. > > > > On Wednesday 09 January 2013 15:33:57 Ken Rice wrote: > >> If you are trying to filter NANPA numbers why not use a tighter REGEX > >> like > >> ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6})$ > >> > >> This would leave the base 10 digit number in $1 > >> > >> On 1/9/13 2:00 PM, "DJB International" wrote: > >>> 1800336940 is 10-digits; therefore, it will match (\d{10}), which will > >>> be > >>> $1. > >>> > >>> $1 = 1800336940 > >>> > >>> Then, in your bridge you add 1 in front, so it is correct. > >>> > >>> What behaviors are you looking for in this case? > >>> > >>> -djbinter > >>> > >>> On Wed, Jan 9, 2013 at 11:48 AM, Sean Devoy wrote: > >>>> HI, > >>>> > >>>> I really hope this question is not as dumb as some of my recent > >>>> questions > >>>> (Anybody else wiped out you O/S lately?)! > >>>> > >>>> Anyway, I have an xml dial plan: > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> >>>> > >>>> data="effective_caller_id_number=${outbound_caller_id_number}"/> > >>>> > >>>> >>>> data="ani=${outbound_caller_id_number}"/> > >>>> >>>> > >>>> data="effective_caller_id_name=${outbound_caller_id_name}"/> > >>>> > >>>> > >>>> >>>> > >>>> data="sofia/gateway/vitelity-inbound/1$1"/> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> I thought that would accept either 10 or 11 digit numbers starting with > >>>> 1 > >>>> and bridge to the gateway always using 1 plus the 10 digits. > >>>> > >>>> In fact in my log I can see: > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) > >>>> [ten_or > >>>> eleven_digiti_dial] destination_number(14104207777 > >>>> ) > >>>> =~ /^1?(\d{10})$/ break=on-false > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(ringback=${us-ring}) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(effective_caller_id_number=${outbound_caller_id_number}) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(ani=${outbound_caller_id_number}) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(effective_caller_id_name=${outbound_caller_id_name}) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(hangup_after_bridge=true) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> bridge(sofia/gateway/vitelity-inbound/14104207777 ) > >>>> > >>>> BUT > >>>> It does not work for my customer : > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) > >>>> [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ > >>>> /^1?(\d{10})$/ break=on-false > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >>>> set(ringback=${us-ring}) > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >>>> set(effective_caller_id_number=${outbound_caller_id_number}) > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >>>> set(ani=${outbound_caller_id_number}) > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >>>> set(effective_caller_id_name=${outbound_caller_id_name}) > >>>> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > >>>> set(hangup_after_bridge=true) > >>>> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > >>>> bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA > >>>> LEADING 1 > >>>> So, what am I missing? > >>>> > >>>> Thanks, > >>>> Sean > >>>> > >>>> _______________________________________________________________________ > >>>> __ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > >>>> s > >>>> http://www.freeswitch.org > >>> > >>> ________________________________________________________________________ > >>> _ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org From krice at freeswitch.org Thu Jan 10 01:18:57 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 09 Jan 2013 16:18:57 -0600 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: Message-ID: Correct djbinter... I missed that \ before the +... If you don?t do that you?ll get a nothing to repeat compile error on the regex... Thanks for pointing it out! On 1/9/13 4:03 PM, "DJB International" wrote: > Ken, > > Is it correct that in the group but not capture (?: we will need an escape > character \ to match the literal "+" character as well.? So, I think it should > be: > ^(?:\+1|1)?([2-9]\d{2}[2-9]\d{6})$ > > -djbinter > > > > On Wed, Jan 9, 2013 at 1:44 PM, Sergey Okhapkin > wrote: >> It's better to remove trailing "$", there could be more digits on handset's >> redial after entering DTMF codes on previous call. >> >> On Wednesday 09 January 2013 15:33:57 Ken Rice wrote: >>> > If you are trying to filter NANPA numbers why not use a tighter REGEX like >>> > ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6})$ >>> > >>> > This would leave the base 10 digit number in $1 >>> > >>> > On 1/9/13 2:00 PM, "DJB International" wrote: >>>> > > 1800336940 is 10-digits; therefore, it will match (\d{10}), which will be >>>> > > $1. >>>> > > >>>> > > $1 = 1800336940 >>>> > > >>>> > > Then, in your bridge you add 1 in front, so it is correct. >>>> > > >>>> > > What behaviors are you looking for in this case? >>>> > > >>>> > > -djbinter >>>> > > >>>> > > On Wed, Jan 9, 2013 at 11:48 AM, Sean Devoy >>>> wrote: >>>>> > >> HI, >>>>> > >> >>>>> > >> I really hope this question is not as dumb as some of my recent >>>>> questions >>>>> > >> (Anybody else wiped out you O/S lately?)! >>>>> > >> >>>>> > >> Anyway, ?I have an xml dial plan: >>>>> > >> ? >>>>> > >> ? ? >>>>> > >> ? ? ? >>>>> > >> ? ? ? >>>> > >> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>>>> > >> ? ? ? >>>> data="ani=${outbound_caller_id_number}"/> >>>>> > >> ? ? ? >>>> > >> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>>>> > >> ? ? ? >>>>> > >> ? ? ? >>>> > >> data="sofia/gateway/vitelity-inbound/1$1"/> >>>>> > >> ? ? >>>>> > >> ? >>>>> > >> >>>>> > >> I thought that would accept either 10 or 11 digit numbers starting >>>>> with 1 >>>>> > >> and bridge to the gateway always using 1 plus the 10 digits. >>>>> > >> >>>>> > >> In fact in my log I can see: >>>>> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) >>>>> [ten_or >>>>> > >> eleven_digiti_dial] destination_number(14104207777 >>>>> ) >>>>> > >> =~ /^1?(\d{10})$/ break=on-false >>>>> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>>> > >> set(ringback=${us-ring}) >>>>> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>>> > >> set(effective_caller_id_number=${outbound_caller_id_number}) >>>>> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>>> > >> set(ani=${outbound_caller_id_number}) >>>>> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>>> > >> set(effective_caller_id_name=${outbound_caller_id_name}) >>>>> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>>> > >> set(hangup_after_bridge=true) >>>>> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>>> > >> bridge(sofia/gateway/vitelity-inbound/14104207777 >>>>> > ) >>>>> > >> >>>>> > >> BUT >>>>> > >> It does not work for my customer : >>>>> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) >>>>> > >> [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ >>>>> > >> /^1?(\d{10})$/ break=on-false >>>>> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >>>>> > >> set(ringback=${us-ring}) >>>>> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >>>>> > >> set(effective_caller_id_number=${outbound_caller_id_number}) >>>>> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >>>>> > >> set(ani=${outbound_caller_id_number}) >>>>> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >>>>> > >> set(effective_caller_id_name=${outbound_caller_id_name}) >>>>> > >> Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action >>>>> > >> set(hangup_after_bridge=true) >>>>> > >> Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action >>>>> > >> bridge(sofia/gateway/vitelity-inbound/11800336940) ?NOTE THE EXTRA >>>>> > >> LEADING 1 >>>>> > >> So, what am I missing? >>>>> > >> >>>>> > >> Thanks, >>>>> > >> Sean >>>>> > >> >>>>> > >> >>>>> _________________________________________________________________________ >>>>> > >> Professional FreeSWITCH Consulting Services: >>>>> > >> consulting at freeswitch.org >>>>> > >> http://www.freeswitchsolutions.com >>>>> > >> >>>>> > >> >>>>> > >> >>>>> > >> >>>>> > >> Official FreeSWITCH Sites >>>>> > >> http://www.freeswitch.org >>>>> > >> http://wiki.freeswitch.org >>>>> > >> http://www.cluecon.com >>>>> > >> >>>>> > >> FreeSWITCH-users mailing list >>>>> > >> FreeSWITCH-users at lists.freeswitch.org >>>>> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > >> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > >> http://www.freeswitch.org >>>> > > >>>> > > >>>> _________________________________________________________________________ >>>> > > Professional FreeSWITCH Consulting Services: >>>> > > consulting at freeswitch.org >>>> > > http://www.freeswitchsolutions.com >>>> > > >>>> > > >>>> > > >>>> > > >>>> > > Official FreeSWITCH Sites >>>> > > http://www.freeswitch.org >>>> > > http://wiki.freeswitch.org >>>> > > http://www.cluecon.com >>>> > > >>>> > > FreeSWITCH-users mailing list >>>> > > FreeSWITCH-users at lists.freeswitch.org >>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > > >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/b2ed5950/attachment.html From schoch+freeswitch.org at xwin32.com Thu Jan 10 01:48:05 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 9 Jan 2013 14:48:05 -0800 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: References: <50DE75C4.1030500@communicatefreely.net> <1ADA846F-0D19-4FE3-AFA3-7CA5E8E4C127@insensate.co.uk> <2ef401cdee82$090f9df0$1b2ed9d0$@bizfocused.com> Message-ID: On Wed, Jan 9, 2013 at 1:27 PM, Lawrence Conroy wrote: > IMHO, most DNS providers are the sticking point -- they wouldn't know a > NAPTR if it jumped up and bit them on the t*t, and have less desire to > change their customer dashboard than pretty much anything; if it works for > customer web sites, why would they need to do anything else? Case in point: On Wed, Jan 9, 2013 at 12:08 PM, QuickServe Support wrote: > Hello Steve, > > > The "Add New Custom DNS Entries" page does not have a "NAPTR" entry > > (as specified in RFC 2915). Is this something that can be added? > > Unfortunately, the NAPTR DNS record is not supported by our systems at this > time. Currently there are no plans to add support for the NAPTR DNS > record. > > I'm not aware of any third party DNS provide that does offer NAPTR DNS to > direct you to. > > > Best regards, > Sean P. > pair Networks, Inc. > support at pair.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/b35e2296/attachment-0001.html From sdevoy at bizfocused.com Thu Jan 10 01:53:16 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 9 Jan 2013 17:53:16 -0500 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> Message-ID: <33d801cdeebc$1d104400$5730cc00$@bizfocused.com> Hey Avi, Using your regex: ^(?:1|\+1|001)?([2-9]\d\d[2-9]\d{6})$ Doesn't $1 match the stuff in the first set of () and $2 match my needed 10 digits? Thanks, Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Wednesday, January 09, 2013 3:02 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! Ha, that was a tricky one! Here's the issue - he dialed: 1-800-336-940 -- that's too few digits! So your optional one made it match anyway. Your matching is too allowing - here's the regex I use: ^(?:1|\+1|001)?([2-9]\d\d[2-9]\d{6})$ That matches [nothing, 1, +1 or 001] prefix for USA. Then, bridge like now to "1$1". btw, if you want a separate match for USA toll-free it's: ^(?:1|\+1|001)?(8(00|55|66|77|88)[2-9]\d{6})$ There's also some good example on this page, the link goes directly to USA: http://wiki.freeswitch.org/wiki/Regular_Expression#NANPA_.2B1NxxNxxXXXX_E.16 4_Dialstring -Avi Marcus BestFone On Wed, Jan 9, 2013 at 9:48 PM, Sean Devoy wrote: HI, I really hope this question is not as dumb as some of my recent questions (Anybody else wiped out you O/S lately?)! Anyway, I have an xml dial plan: I thought that would accept either 10 or 11 digit numbers starting with 1 and bridge to the gateway always using 1 plus the 10 digits. In fact in my log I can see: Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) [ten_or eleven_digiti_dial] destination_number(14104207777) =~ /^1?(\d{10})$/ break=on-false Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(ringback=${us-ring}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(ani=${outbound_caller_id_number}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(hangup_after_bridge=true) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action bridge(sofia/gateway/vitelity-inbound/14104207777) BUT It does not work for my customer : Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ /^1?(\d{10})$/ break=on-false Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(ringback=${us-ring}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(ani=${outbound_caller_id_number}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(hangup_after_bridge=true) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA LEADING 1 So, what am I missing? Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/8fdd1335/attachment.html From sdevoy at bizfocused.com Thu Jan 10 01:54:37 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 9 Jan 2013 17:54:37 -0500 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> Message-ID: <33e601cdeebc$4d2ab800$e7802800$@bizfocused.com> Sorry, disregard my question about $1 vs $2 - I re-read the wiki. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Wednesday, January 09, 2013 3:02 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! Ha, that was a tricky one! Here's the issue - he dialed: 1-800-336-940 -- that's too few digits! So your optional one made it match anyway. Your matching is too allowing - here's the regex I use: ^(?:1|\+1|001)?([2-9]\d\d[2-9]\d{6})$ That matches [nothing, 1, +1 or 001] prefix for USA. Then, bridge like now to "1$1". btw, if you want a separate match for USA toll-free it's: ^(?:1|\+1|001)?(8(00|55|66|77|88)[2-9]\d{6})$ There's also some good example on this page, the link goes directly to USA: http://wiki.freeswitch.org/wiki/Regular_Expression#NANPA_.2B1NxxNxxXXXX_E.16 4_Dialstring -Avi Marcus BestFone On Wed, Jan 9, 2013 at 9:48 PM, Sean Devoy wrote: HI, I really hope this question is not as dumb as some of my recent questions (Anybody else wiped out you O/S lately?)! Anyway, I have an xml dial plan: I thought that would accept either 10 or 11 digit numbers starting with 1 and bridge to the gateway always using 1 plus the 10 digits. In fact in my log I can see: Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex (PASS) [ten_or eleven_digiti_dial] destination_number(14104207777) =~ /^1?(\d{10})$/ break=on-false Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(ringback=${us-ring}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(ani=${outbound_caller_id_number}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(hangup_after_bridge=true) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action bridge(sofia/gateway/vitelity-inbound/14104207777) BUT It does not work for my customer : Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex (PASS) [ten_or_eleven_digiti_dial] destination_number(1800336940) =~ /^1?(\d{10})$/ break=on-false Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(ringback=${us-ring}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(ani=${outbound_caller_id_number}) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action set(hangup_after_bridge=true) Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action bridge(sofia/gateway/vitelity-inbound/11800336940) NOTE THE EXTRA LEADING 1 So, what am I missing? Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/908f5305/attachment-0001.html From avi at avimarcus.net Thu Jan 10 01:59:23 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 10 Jan 2013 00:59:23 +0200 Subject: [Freeswitch-users] xml dialplan - regurlar expression question / Sanity Check! In-Reply-To: <33d801cdeebc$1d104400$5730cc00$@bizfocused.com> References: <320801cdeea2$499ef800$dcdce800$@bizfocused.com> <33d801cdeebc$1d104400$5730cc00$@bizfocused.com> Message-ID: ?: means "match but don't save it", for when [] just isn't flexible enough. -Avi On Thu, Jan 10, 2013 at 12:53 AM, Sean Devoy wrote: > Hey Avi,**** > > ** ** > > Using your regex: ^(?:1|\+1|001)?([2-9]\d\d[2-9]\d{6})$**** > > Doesn?t $1 match the stuff in the first set of () and $2 match my needed > 10 digits?**** > > ** ** > > Thanks,**** > > Sean**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Wednesday, January 09, 2013 3:02 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] xml dialplan - regurlar expression > question / Sanity Check!**** > > ** ** > > Ha, that was a tricky one!**** > > ** ** > > Here's the issue - he dialed: 1-800-336-940 -- that's too few digits!**** > > So your optional one made it match anyway.**** > > ** ** > > Your matching is too allowing - here's the regex I use:**** > > ** ** > > ^(?:1|\+1|001)?([2-9]\d\d[2-9]\d{6})$**** > > ** ** > > That matches [nothing, 1, +1 or 001] prefix for USA. Then, bridge like now > to "1$1".**** > > ** ** > > btw, if you want a separate match for USA toll-free > it's: ^(?:1|\+1|001)?(8(00|55|66|77|88)[2-9]\d{6})$**** > > ** ** > > There's also some good example on this page, the link goes directly to > USA: > http://wiki.freeswitch.org/wiki/Regular_Expression#NANPA_.2B1NxxNxxXXXX_E.164_Dialstring > **** > > > **** > > -Avi Marcus > BestFone**** > > ** ** > > On Wed, Jan 9, 2013 at 9:48 PM, Sean Devoy wrote:* > *** > > HI,**** > > **** > > I really hope this question is not as dumb as some of my recent questions > (Anybody else wiped out you O/S lately?)!**** > > **** > > Anyway, I have an xml dial plan:**** > > **** > > **** > > **** > > **** > > * > *** > > **** > > **** > > **** > > **** > > **** > > **** > > I thought that would accept either 10 or 11 digit numbers starting with 1 > and bridge to the gateway always using 1 plus the 10 digits.**** > > **** > > In fact in my log I can see:**** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Regex *(PASS)*[ten_or eleven_digiti_dial] destination_number > *(14104207777*) =~ /^1?(\d{10})$/ break=on-false**** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ringback=${us-ring}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(ani=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action bridge* > (sofia/gateway/vitelity-inbound/14104207777*)**** > > **** > > *BUT***** > > It does not work for my customer :**** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Regex *(PASS*) > [ten_or_eleven_digiti_dial] destination_number(*1800336940*) =~ > /^1?(\d{10})$/ break=on-false**** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ringback=${us-ring}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_number=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(ani=${outbound_caller_id_number}) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action > set(effective_caller_id_name=${outbound_caller_id_name}) **** > > Dialplan: sofia/external/220 at fs_bfis.bizfocused.com Action > set(hangup_after_bridge=true) **** > > Dialplan: sofia/external/101 at fs_alst.bizfocused.com Action bridge* > (sofia/gateway/vitelity-inbound/11800336940*) NOTE THE EXTRA LEADING 1*** > * > > **** > > So, what am I missing?**** > > **** > > Thanks,**** > > Sean**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/af08220a/attachment.html From dujinfang at gmail.com Thu Jan 10 02:26:33 2013 From: dujinfang at gmail.com (Seven Du) Date: Thu, 10 Jan 2013 07:26:33 +0800 Subject: [Freeswitch-users] Issue: Invalid codec mpeg4-generic! In-Reply-To: References: Message-ID: it would be easy to convert to H264 with ffmpeg or handbrake -- Seven Du Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Wednesday, January 9, 2013 at 9:33 AM, Muqtheear S wrote: > mod_mp4 module is looking for Hint tracks in the .mp4 file. "Hint tracks are a series of instructions in a mp4 container file that tell a server how to transmit packets." To create hint tracks for a mp4 file is fairly simple with the mp4creator program, we just executed the: mp4creator -hint= [-p ] As we know .mp4 has both video+audio data in it. We could also find audio and video encoding information of a .mp4 file as shown below. #mp4info minnale.mp4 mp4info version 1.5.0.1 minnale.mp4: Track Type Info 1 video MPEG-4 Simple @ L1, 97.833 secs, 136 kbps, 176x144 @ 12.000041 fps 2 audio MPEG-4 AAC LC, 97.568 secs, 64 kbps, 32000 Hz 3 hint Payload MP4V-ES for track 1 4 hint Payload mpeg4-generic for track 2 Metadata Tool: mp4creator 1.6 Now we are getting "Invalid codec mpeg4-generic!" error. Please guide me. Do we need to have mpeg4-generic codec in Freeswitch server? > > > Thanks & Regards, > Muqtheear.S > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/9caff287/attachment-0001.html From dujinfang at gmail.com Thu Jan 10 02:29:30 2013 From: dujinfang at gmail.com (Seven Du) Date: Thu, 10 Jan 2013 07:29:30 +0800 Subject: [Freeswitch-users] Issue: Invalid codec mpeg4-generic! In-Reply-To: References: Message-ID: <56A98C666D5244018C5620B890263D30@gmail.com> Sorry, for the sound track, I guess you need convert to PCMU, if I remember right, the last time I look the code it said only PCMU is supported. -- Seven Du Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, January 10, 2013 at 7:26 AM, Seven Du wrote: > it would be easy to convert to H264 with ffmpeg or handbrake > > -- > Seven Du > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > On Wednesday, January 9, 2013 at 9:33 AM, Muqtheear S wrote: > > > mod_mp4 module is looking for Hint tracks in the .mp4 file. "Hint tracks are a series of instructions in a mp4 container file that tell a server how to transmit packets." To create hint tracks for a mp4 file is fairly simple with the mp4creator program, we just executed the: mp4creator -hint= [-p ] As we know .mp4 has both video+audio data in it. We could also find audio and video encoding information of a .mp4 file as shown below. #mp4info minnale.mp4 mp4info version 1.5.0.1 minnale.mp4: Track Type Info 1 video MPEG-4 Simple @ L1, 97.833 secs, 136 kbps, 176x144 @ 12.000041 fps 2 audio MPEG-4 AAC LC, 97.568 secs, 64 kbps, 32000 Hz 3 hint Payload MP4V-ES for track 1 4 hint Payload mpeg4-generic for track 2 Metadata Tool: mp4creator 1.6 Now we are getting "Invalid codec mpeg4-generic!" error. Please guide me. Do we need to have mpeg4-generic codec in Freeswitch server? > > > > > > Thanks & Regards, > > Muqtheear.S > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/26c70cb3/attachment.html From 8f27e956 at gmail.com Thu Jan 10 02:31:23 2013 From: 8f27e956 at gmail.com (S. Scott) Date: Wed, 9 Jan 2013 18:31:23 -0500 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: References: <50DE75C4.1030500@communicatefreely.net> <1ADA846F-0D19-4FE3-AFA3-7CA5E8E4C127@insensate.co.uk> <2ef401cdee82$090f9df0$1b2ed9d0$@bizfocused.com> Message-ID: <6884998629468252395@unknownmsgid> www.zoneedit.com Enrollment is free. Depending on number of domains and forward zones requirements, you may even remain within the for-free thresholds. /S ????? iThing: Big thumbs & little keys. Please excuse typo, spelling and grammar errors ? Thought of the Day ? "With all this manure, there must be a pony in here somewhere.? On 2013-01-09, at 17:52, Steven Schoch wrote: On Wed, Jan 9, 2013 at 1:27 PM, Lawrence Conroy wrote: > IMHO, most DNS providers are the sticking point -- they wouldn't know a > NAPTR if it jumped up and bit them on the t*t, and have less desire to > change their customer dashboard than pretty much anything; if it works for > customer web sites, why would they need to do anything else? Case in point: On Wed, Jan 9, 2013 at 12:08 PM, QuickServe Support wrote: > Hello Steve, > > > The "Add New Custom DNS Entries" page does not have a "NAPTR" entry > > (as specified in RFC 2915). Is this something that can be added? > > Unfortunately, the NAPTR DNS record is not supported by our systems at this > time. Currently there are no plans to add support for the NAPTR DNS > record. > > I'm not aware of any third party DNS provide that does offer NAPTR DNS to > direct you to. > > > Best regards, > Sean P. > pair Networks, Inc. > support at pair.com > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/92c1a5a9/attachment.html From 8f27e956 at gmail.com Thu Jan 10 02:35:31 2013 From: 8f27e956 at gmail.com (S. Scott) Date: Wed, 9 Jan 2013 18:35:31 -0500 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: References: <50DE75C4.1030500@communicatefreely.net> <1ADA846F-0D19-4FE3-AFA3-7CA5E8E4C127@insensate.co.uk> <2ef401cdee82$090f9df0$1b2ed9d0$@bizfocused.com> Message-ID: <4090664200245465731@unknownmsgid> Also, on the paid dns provider front, easydns.com, a VERY good dns provider, supports the full monty of record types, http://helpwiki.easydns.com/index.php/Editing_NAPTR_record /S ????? iThing: Big thumbs & little keys. Please excuse typo, spelling and grammar errors ? Thought of the Day ? "With all this manure, there must be a pony in here somewhere.? On 2013-01-09, at 17:52, Steven Schoch wrote: On Wed, Jan 9, 2013 at 1:27 PM, Lawrence Conroy wrote: > IMHO, most DNS providers are the sticking point -- they wouldn't know a > NAPTR if it jumped up and bit them on the t*t, and have less desire to > change their customer dashboard than pretty much anything; if it works for > customer web sites, why would they need to do anything else? Case in point: On Wed, Jan 9, 2013 at 12:08 PM, QuickServe Support wrote: > Hello Steve, > > > The "Add New Custom DNS Entries" page does not have a "NAPTR" entry > > (as specified in RFC 2915). Is this something that can be added? > > Unfortunately, the NAPTR DNS record is not supported by our systems at this > time. Currently there are no plans to add support for the NAPTR DNS > record. > > I'm not aware of any third party DNS provide that does offer NAPTR DNS to > direct you to. > > > Best regards, > Sean P. > pair Networks, Inc. > support at pair.com > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/1f7bfd16/attachment.html From phillipwarner at gmail.com Thu Jan 10 02:41:12 2013 From: phillipwarner at gmail.com (Phillip Warner) Date: Wed, 9 Jan 2013 15:41:12 -0800 Subject: [Freeswitch-users] mod_directory menu-top? Message-ID: <36C9DE1E-90D7-4827-A218-A012D2D04C76@gmail.com> Hi, is there a parameter in mod_directory to have it transfer (back) to an ivr if the user decides not to search by directory instead of having to hang-up and call again? For example: ivr --> directory --> user changes mind about dialling by name and wants to return to ivr --> ivr Thanks. Phil. From cmason at frontiernetworks.ca Thu Jan 10 10:09:43 2013 From: cmason at frontiernetworks.ca (Colin Mason) Date: Thu, 10 Jan 2013 02:09:43 -0500 Subject: [Freeswitch-users] mod_com_g729 transcoding Message-ID: <0D1C698866F66045A6201FD0F59CAC900146FB2BAB@EX.frontier.local> I have phone A connected to freeswitch box 1 and phone B connected to freeswitch box 2. Phone A wants to dial phone B using G729. Codec is always G729. The path RTP follows is: Phone A -------> FreeSWITCH 1 -------> FreeSWITCH 2 -------> Phone B (g729) (g729) (g729) My question is, why is it that the freeswitch box receiving the call uses up an encoder/decoder when the codec is G729 along the path? If I reverse the call and call Phone A from Phone B, FreeSWITCH Box 1 uses up 1 license. if I dial the PSTN to a carrier who supports G729, I don't use up a license. Any thoughts? Maybe this is normal. FreeSWITCH Box 1: 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP [sofia/mpls/I888_1 at 172.17.17.17] 172.17.17.17 port 32014 -> 10.253.200.6 port 16466 codec: 18 ms: 20 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP [sofia/transport/2996] 172.17.17.17 port 27276 -> 172.16.16.16 port 17056 codec: 18 ms: 20 freeswitch at internal> g729_info Permitted G729 channels: 40 Encoders in use: 0 Decoders in use: 0 FreeSWITCH Box 2: 2013-01-10 01:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP [sofia/transport/3888 at 172.17.17.17] 172.16.16.16 port 17056 -> 172.17.17.17 port 27276 codec: 18 ms: 20 2013-01-10 01:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP [sofia/mpls/sip:C996_1 at 10.253.200.10:5060] 172.16.16.16 port 18966 -> 10.253.200.10 port 16486 codec: 18 ms: 20 freeswitch at internal> g729_info Permitted G729 channels: 40 Encoders in use: 1 Decoders in use: 1 Thanks in advance guys. Colin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/8a7444a0/attachment-0001.html From muqtheear.s at gmail.com Thu Jan 10 10:31:37 2013 From: muqtheear.s at gmail.com (Muqtheear S) Date: Thu, 10 Jan 2013 13:01:37 +0530 Subject: [Freeswitch-users] Issue: Invalid codec mpeg4-generic! In-Reply-To: <56A98C666D5244018C5620B890263D30@gmail.com> References: <56A98C666D5244018C5620B890263D30@gmail.com> Message-ID: Thanks for the reply. 1. Please tell me how to convert mpeg4-generic encoded audio data to PCMU encoded audio data in .mp4 file. I tried with ffmpeg, but couldn't succeed. 2. If you have any mp4 file with PCMU sound track, please share with me. 3. Does freeswitch comes with mpeg4-generic codec support? Thanks & Regards, Muqtheear.S On Thu, Jan 10, 2013 at 4:59 AM, Seven Du wrote: > Sorry, for the sound track, I guess you need convert to PCMU, if I > remember right, the last time I look the code it said only PCMU is > supported. > > -- > Seven Du > Sent with Sparrow > > On Thursday, January 10, 2013 at 7:26 AM, Seven Du wrote: > > it would be easy to convert to H264 with ffmpeg or handbrake > > -- > Seven Du > Sent with Sparrow > > On Wednesday, January 9, 2013 at 9:33 AM, Muqtheear S wrote: > > mod_mp4 module is looking for Hint tracks in the .mp4 file. > "Hint tracks are a series of instructions in a mp4 container file that > tell a server how to transmit packets." > To create hint tracks for a mp4 file is fairly simple with the mp4creator > program, we just executed the: > mp4creator -hint= [-p ] > > As we know .mp4 has both video+audio data in it. We could also find audio > and video encoding information of a .mp4 file as shown below. > > #mp4info minnale.mp4 > mp4info version 1.5.0.1 > minnale.mp4: > Track Type Info > 1 video MPEG-4 Simple @ L1, 97.833 secs, 136 kbps, 176x144 @ > 12.000041 fps > 2 audio MPEG-4 AAC LC, 97.568 secs, 64 kbps, 32000 Hz > 3 hint Payload MP4V-ES for track 1 > 4 hint Payload mpeg4-generic for track 2 > Metadata Tool: mp4creator 1.6 > Now we are getting "Invalid codec mpeg4-generic!" error. > Please guide me. Do we need to have mpeg4-generic codec in Freeswitch server? > > > Thanks & Regards, > Muqtheear.S > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/5103bda3/attachment.html From wangd at alongtechnology.com.cn Thu Jan 10 04:39:22 2013 From: wangd at alongtechnology.com.cn (=?gb2312?B?zfW1zw==?=) Date: Thu, 10 Jan 2013 09:39:22 +0800 Subject: [Freeswitch-users] How to pause and resume play an audio file in conference? Message-ID: <201301100939217208404@alongtechnology.com.cn> How to pause and resume play an audio file in conference? For example: I create a conference room with some members, and play an audio file to them. When someone start-talking, I pause the playing. When nobody talking, I resume play the audio file. BTW, In freeswitch-1.2rc, I tryed play,pause,resume commands, but pause and resume is for recording, not for my needs. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/a41877e9/attachment.html From craig.robert at hotmail.com Thu Jan 10 06:42:40 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Wed, 9 Jan 2013 20:42:40 -0700 Subject: [Freeswitch-users] T38 Configuration Message-ID: Just installed FusionPBX using the script built for a Centos 6 install. Everything works great in regards to calls, but my main goal of this project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have tried many things in media bypass, routing to a fax extension, using an ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me in the right direction as to where I should begin? I am guessing it isn't supposed to work right out of the box. I am using Flowroute as my provider right now and the have confirmed that they utilize T38 in both directions. Thanks! Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130109/426ff182/attachment.html From andrew at cassidywebservices.co.uk Thu Jan 10 11:54:50 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 10 Jan 2013 08:54:50 +0000 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: <4090664200245465731@unknownmsgid> References: <50DE75C4.1030500@communicatefreely.net> <1ADA846F-0D19-4FE3-AFA3-7CA5E8E4C127@insensate.co.uk> <2ef401cdee82$090f9df0$1b2ed9d0$@bizfocused.com> <4090664200245465731@unknownmsgid> Message-ID: https://dns.he.net/ Free to an extent but I can't remember what the limits are. Something like 50 zones. On 9 January 2013 23:35, S. Scott <8f27e956 at gmail.com> wrote: > Also, on the paid dns provider front, easydns.com, a VERY good dns > provider, supports the full monty of record types, > > http://helpwiki.easydns.com/index.php/Editing_NAPTR_record > > /S > > ????? > iThing: Big thumbs & little keys. Please excuse typo, spelling and > grammar errors ? Thought of the Day ? "With all this manure, there must be > a pony in here somewhere.? > > On 2013-01-09, at 17:52, Steven Schoch > wrote: > > On Wed, Jan 9, 2013 at 1:27 PM, Lawrence Conroy wrote: > >> IMHO, most DNS providers are the sticking point -- they wouldn't know a >> NAPTR if it jumped up and bit them on the t*t, and have less desire to >> change their customer dashboard than pretty much anything; if it works for >> customer web sites, why would they need to do anything else? > > > Case in point: > > On Wed, Jan 9, 2013 at 12:08 PM, QuickServe Support wrote: > >> Hello Steve, >> >> > The "Add New Custom DNS Entries" page does not have a "NAPTR" entry >> > (as specified in RFC 2915). Is this something that can be added? >> >> Unfortunately, the NAPTR DNS record is not supported by our systems at >> this >> time. Currently there are no plans to add support for the NAPTR DNS >> record. >> >> I'm not aware of any third party DNS provide that does offer NAPTR DNS to >> direct you to. >> >> >> Best regards, >> Sean P. >> pair Networks, Inc. >> support at pair.com >> > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/959732ba/attachment-0001.html From avi at avimarcus.net Thu Jan 10 12:31:49 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 10 Jan 2013 11:31:49 +0200 Subject: [Freeswitch-users] How to pause and resume play an audio file in conference? In-Reply-To: <201301100939217208404@alongtechnology.com.cn> References: <201301100939217208404@alongtechnology.com.cn> Message-ID: It looks like you can do it via api command: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman It seems the conference has stop-talking and start-talking events, so you can start the file and then have an ESL app that listens for the events and stops/starts the playback. Let us know how that goes. -Avi On Thu, Jan 10, 2013 at 3:39 AM, ?? wrote: > ** > How to pause and resume play an audio file in conference? For example: > I create a conference room with some members, and play an audio file to > them. When someone start-talking, I pause the playing. When nobody talking, > I resume play the audio file. > > BTW, In freeswitch-1.2rc, I tryed play,pause,resume commands, but pause > and resume is for recording, not for my needs. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/ccaaa4ee/attachment.html From steveayre at gmail.com Thu Jan 10 12:31:29 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 10 Jan 2013 09:31:29 +0000 Subject: [Freeswitch-users] Best practices question about SIP registration In-Reply-To: <7D1B08FC-5E9C-4E58-A9BC-B68968EB0187@insensate.co.uk> References: <50DE75C4.1030500@communicatefreely.net> <7D1B08FC-5E9C-4E58-A9BC-B68968EB0187@insensate.co.uk> Message-ID: > There IS a wrinkle to this special case: If you have a URL of the form the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host The reason being that a SRV record includes the port (note that means it's a very easy way to add/change additional ports as well as addresses without having to change customer configs). So obviously a host:port isn't compatible with a SRV, since it wouldn't make sense to override the port, so the assumption is that it's specifying an A/AAAA record instead and port on that IP instead. -Steve On 8 January 2013 23:35, Lawrence Conroy wrote: > Hi Michael, folks, > if you're going to codify it, I did slightly simplify things: > (full disclosure -- I disagreed with it at the time which is probably why I forgot to mention it -- honest). > > There IS a wrinkle to this special case: If you have a URL of the form the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host example.com (i.e., there's a machine called example.com, and it is handling SIP traffic for that domainpart). > Basically, if you see a colon in the domainpart, you're looking for a machine -- otherwise you're looking for a NAPTR (and/or a SRV at _sip._udp.). > > I'd put that before the paragraph starting "However, relying on ..." > > Curiously enough, the old 2543-compliant servers did hunt the SRV rather than giving up and looking for an A, so this was a change. Such fun was had re-writing implementations (plural) and testing them yet again. Sigh. > > all the best, > Lawrence > > > On 8 Jan 2013, at 22:56, Michael Collins wrote: > >> Lawrence, >> >> Thanks for this explanation. It was very well written. I'm looking for a >> place to codify this on the wiki so that it gets preserved... :) >> >> -MC >> >> On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy wrote: >> >>> Hi there, >>> at the risk of butting in on someone else's party ... >>> Nope; your interpretations is NOT best practice. >>> I have some sympathy, as the term domain is overloaded within fS. >>> >>> A sip address consists of a userpart and a domain part -- e.g., >>> >>> The sip domain is similar to an email domain -- e.g., >> user at maildomain> >>> With email, you need to do a lookup of the MX record in DNS to find the >>> FQDN of the machine that handles mail for the domain. >>> With SIP (see RFC 3263), you do a lookup on the SRV record (at >>> _sip._udp.) to find the machine that handles SIP >>> registrations/incalls for the domain. That also gives you the port on >>> which that machine is listening. >>> (Yup, you can also have a NAPTR record in the domain to tell you where the >>> SRV record is, but many folks don't bother -- for Best Practice, you >>> should, but ...) >>> >>> There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) >>> support that means most SIP clients will look for the SRV and, if it can't >>> be found (or there's an IP address rather than a DNS -style domain, in >>> which case the SIP client won't bother hunting the SRV), the client will >>> guess that the domain has a machine (i.e. it will look for an A or AAAA >>> record), and also guess it's listening on 5060 (the default port). >>> Email is the same (mail to fred at example.com, and strictly the sender will >>> do a check for a MX and then look for an A record for example.com, and >>> try there). >>> >>> However, relying on that default "get out" clause is definitely NOT what >>> you should do for BCP. >>> Using the hostname as the sip domain is a kludge -- the FQDN with A record >>> usually works, but it's not what you want to do. >>> >>> SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at >>> _sip._udp., and you're done. No need for an A record at that domain >>> at all. >>> >>> (RFC 3263 is not too hard to read, for a change -- it's certainly shorter >>> than RFC 3261, and it even has an ASCII art diagram :). >>> >>> all the best, >>> Lawrence >>> >>> On 8 Jan 2013, at 21:05, Steven Schoch wrote: >>> >>>> On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < >>>> fs-list at communicatefreely.net> wrote: >>>> >>>>> Hi Steven, >>>>> >>>>> I would recommend using a proper domain name as much as possible. For >>>>> one, it looks >>>>> nicer! A SIP URI is supposed to be user at domain like an e-mail address >>>>> is, and I hope that >>>>> one day URI dialing will be common place, so we might as well do it >>> right >>>>> the first time. >>>>> >>>> >>>> What you're saying is that "domain" should really be a fully-qualified >>> host >>>> name that points via DNS to the actual host on which FreeSwitch is >>> running. >>>> That is, the domain should be "pbx.example.com" instead of just " >>>> example.com", as the last example would most likely point to a web >>> server, >>>> not the SIP server. Do I have that right? >>>> >>>> Next, in the configuration for Polycom phones (for example), there are 2 >>>> fields that both have the userid. In the example in >>>> http://wiki.freeswitch.org/wiki/Polycom_configuration it has: >>>> >>>> reg.1.auth.userId="1000" >>>> >>>> and >>>> >>>> reg.1.address="1000 at fs.domain.local" >>>> >>>> How is the "address" value used? Is that sent in the SIP registration >>>> message? If that's the case, what does Freeswitch do with it? >>>> >>>> -- >>>> Steve >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveayre at gmail.com Thu Jan 10 12:38:52 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 10 Jan 2013 09:38:52 +0000 Subject: [Freeswitch-users] mod_com_g729 transcoding In-Reply-To: <0D1C698866F66045A6201FD0F59CAC900146FB2BAB@EX.frontier.local> References: <0D1C698866F66045A6201FD0F59CAC900146FB2BAB@EX.frontier.local> Message-ID: A more complete debug-level log could be useful to get more context. Is box 2 running any other calls at the same time that might be using the license? Is box 2 doing anything else with the call? Anything like eavesdrop, recording, start_dtmf / start_dtmf_generate etc which uses the media will use a license, and I *think* that license only gets released until the end of the call. -Steve On 10 January 2013 07:09, Colin Mason wrote: > I have phone A connected to freeswitch box 1 and phone B connected to > freeswitch box 2. > > > > Phone A wants to dial phone B using G729. Codec is always G729. > > The path RTP follows is: > > > > > > > > Phone A -------> FreeSWITCH 1 -------> FreeSWITCH 2 -------> Phone B > > (g729) (g729) > (g729) > > > > > > > > My question is, why is it that the freeswitch box receiving the call uses up > an encoder/decoder when the codec is G729 along the path? If I reverse the > call and call Phone A from Phone B, FreeSWITCH Box 1 uses up 1 license. > > if I dial the PSTN to a carrier who supports G729, I don?t use up a license. > Any thoughts? Maybe this is normal. > > > > FreeSWITCH Box 1: > > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP > [sofia/mpls/I888_1 at 172.17.17.17] 172.17.17.17 port 32014 -> 10.253.200.6 > port 16466 codec: 18 ms: 20 > > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP > [sofia/transport/2996] 172.17.17.17 port 27276 -> 172.16.16.16 port 17056 > codec: 18 ms: 20 > > freeswitch at internal> g729_info > > Permitted G729 channels: 40 > > Encoders in use: 0 > > Decoders in use: 0 > > > > FreeSWITCH Box 2: > > 2013-01-10 01:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP > [sofia/transport/3888 at 172.17.17.17] 172.16.16.16 port 17056 -> 172.17.17.17 > port 27276 codec: 18 ms: 20 > > 2013-01-10 01:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP > [sofia/mpls/sip:C996_1 at 10.253.200.10:5060] 172.16.16.16 port 18966 -> > 10.253.200.10 port 16486 codec: 18 ms: 20 > > freeswitch at internal> g729_info > > Permitted G729 channels: 40 > > Encoders in use: 1 > > Decoders in use: 1 > > > > > > > > Thanks in advance guys. > > Colin > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From cstomi.levlist at gmail.com Thu Jan 10 14:24:00 2013 From: cstomi.levlist at gmail.com (Tamas.Cseke ) Date: Thu, 10 Jan 2013 12:24:00 +0100 Subject: [Freeswitch-users] Early media without bridge Message-ID: <50EEA4D0.4020208@gmail.com> Hello, We would like to hear early media without CHANNEL_BRIDGE event These are failed calls and callers would like to hear the message that the provider plays Because the caller thinks the call is answered if originate returns. as far as I understand: -early media makes the originate return -if we ignore early media the bridge won't return, but we don't hear it we would like both of them, is it possible somehow? I 'm not sure I fully understand all of the ignore_early_media options but I haven't find solution for this, Could you please advise me one, if there is any? I'm thinking about we maybe need a new ignore_early_media option like "consume" but sending the media to the caller instead of dropping it If there isn't already a solution I also would appreciate if you let me know your opinion about this idea Thanks advance, Tamas From jos at firstcom.dk Thu Jan 10 16:36:06 2013 From: jos at firstcom.dk (=?iso-8859-1?Q?Jon_Sch=F8pzinsky?=) Date: Thu, 10 Jan 2013 14:36:06 +0100 Subject: [Freeswitch-users] Loopback Endpoint Message-ID: <111509FCE9EBF74BAB2CB38AAC521D9B23DCDDF44B@helle.dk.hq.firstcom.dk> Hello List, I can see that the "this will destroy the world and this may kill your pets" warning has been removed from the documentation for the Loopback endpoint. Is this an indication that it has become safer to use? I have a specific problem where I essentially have to dial a dialplan for each user I am trying to reach, in parallel. Is it correctly understood that loopback would be the best/only way of implementing this, or is there another way? Kind Regards Jon Sch?pzinsky From abaci64 at gmail.com Thu Jan 10 17:11:09 2013 From: abaci64 at gmail.com (Abaci) Date: Thu, 10 Jan 2013 09:11:09 -0500 Subject: [Freeswitch-users] mod_directory menu-top? In-Reply-To: <36C9DE1E-90D7-4827-A218-A012D2D04C76@gmail.com> References: <36C9DE1E-90D7-4827-A218-A012D2D04C76@gmail.com> Message-ID: <50EECBFC.2030200@gmail.com> use 'execute_extension' to start the directory application so that you get back to the ivr when you exit the directory application. On 1/9/2013 6:41 PM, Phillip Warner wrote: > Hi, is there a parameter in mod_directory to have it transfer (back) to an ivr if the user decides not to search by directory instead of having to hang-up and call again? > > For example: ivr --> directory --> user changes mind about dialling by name and wants to return to ivr --> ivr > > Thanks. Phil. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From michel.brabants at gmail.com Thu Jan 10 17:21:02 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Thu, 10 Jan 2013 15:21:02 +0100 Subject: [Freeswitch-users] freeswitch continously sending sip update-messages: why? In-Reply-To: References: <2D95B81A-2356-43BE-9AC0-A2D0BDC741AB@edvina.net> Message-ID: It indeed was the variable. I was setting it incorrectly ... On Wed, Jan 9, 2013 at 8:13 PM, Kristian Kielhofner wrote: > You are correct. I misread the call flow examples. The ACK was for > the INVITE, not the update. With that being said the correct positive > response to an UPDATE is still a 200, certainly not an ACK (even more > so now). > > On Wed, Jan 9, 2013 at 3:29 AM, Olle E. Johansson wrote: > > > > 8 jan 2013 kl. 21:46 skrev Kristian Kielhofner : > > > >> The correct positive response to an UPDATE would be a 200 that is then > >> ACKd. See RFC 3311: > > The only transaction that needs an ACK is INVITE. > > RFC3311 does not mention an ACK as a response to UPDATE. > > > > /O > > > >> > >> http://tools.ietf.org/html/rfc3311 > >> > >> On Tue, Jan 8, 2013 at 2:40 PM, Steven Ayre > wrote: > >>> Does the client send an ACK for the UPDATE? > >>> > >> > >> > >> -- > >> Kristian Kielhofner > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/223cdee9/attachment.html From victor.chukalovskiy at gmail.com Thu Jan 10 19:03:10 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Thu, 10 Jan 2013 11:03:10 -0500 Subject: [Freeswitch-users] Channel variables for npdi and rn on LRN dipped calls. In-Reply-To: References: Message-ID: <50EEE63E.5090300@gmail.com> Michael and Ken, Thanks for setting me on the right track! What I end-up configuring for now is: ^(?:1)?([2-9]\d{2}[2-9]\d{6})(?:;npdi=(?:yes|no))?(?:;rn=([2-9]\d{2}[2-9]\d{6}))?(?:;npdi=(?:yes|no))? It puts destination number in $1. It also puts LRN in $2 provided it's supplied. Either yes or no are allowed for npdi=. I'm not capturing npdi= into any variable because as I realized all I really need is LRN. For example, all of these: 6045555555;npdi=yes;rn=7785550001 6045555555;rn=7785550001;npdi=yes 16045555555;npdi=yes;rn=7785550001 6045555555;rn=7785550001 Will pass and result in: $1=6045555555 $2=7785550001 And all of these: 6045555555 16045555555 6045555555;npdi=yes 6045555555;npdi=no Will also pass and give: $1=6045555555 $2=empty I'll be glad if this regexp example helps someone. Cheers, -Victor On 13-01-07 08:59 PM, Michael Collins wrote: > In this case the ? modifies the *, so there's .* (greedy) and .*? (not > greedy) > > You want your .* to match as little as possible instead of as much as > possible. If you use .* instead of .*? then your $2 would contain both > the ndpi info but also it would have ";rn=18005551212" b/c the .* > would grab all that stuff. The .*? would only grab up to the ";rn". > > Just adding that info for posterity's sake. :) > > -MC > > On Mon, Jan 7, 2013 at 5:05 PM, Ken Rice > wrote: > > Ooops yeah I mean .*... The ? Is pretty useless there... Since > there should always be something if they send ndpi= > > > On 1/7/13 7:00 PM, "Michael Collins" > wrote: > > > Just a quick FYI... be careful when you use .* because it will > try to match EVERYTHING. Also, Ken accidentally typed (*.) > which I'm sure wasn't what he was aiming for. ;) > > Maybe this would be more effective: > ^(?:+1|1)?([2-9]\d{2}[2-9]\d{6});ndpi=(.*?);rn=([2-9]\d{2}[2-9]\d{6})$ > > -MC > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/8a7761bf/attachment.html From msc at freeswitch.org Thu Jan 10 20:37:50 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Jan 2013 09:37:50 -0800 Subject: [Freeswitch-users] Loopback Endpoint In-Reply-To: <111509FCE9EBF74BAB2CB38AAC521D9B23DCDDF44B@helle.dk.hq.firstcom.dk> References: <111509FCE9EBF74BAB2CB38AAC521D9B23DCDDF44B@helle.dk.hq.firstcom.dk> Message-ID: The devs have done their best to make loopback not be evil but in some respects it must be, because it is what is it. Some folks use it all the time without any issues. Just manage your expectations. :) That being said, you mentioned you need to dial a "dialplan" for each user and that you need to do so in parallel. Without more information I can't offer a better solution, although when I hear things like "dial in parallel" I immediately think "enterprise originate." More information is found on the wikior in the FS Cookbook . -MC On Thu, Jan 10, 2013 at 5:36 AM, Jon Sch?pzinsky wrote: > Hello List, > > I can see that the "this will destroy the world and this may kill your > pets" warning has been removed from the documentation for the Loopback > endpoint. > > Is this an indication that it has become safer to use? > > I have a specific problem where I essentially have to dial a dialplan for > each user I am trying to reach, in parallel. > Is it correctly understood that loopback would be the best/only way of > implementing this, or is there another way? > > > Kind Regards > > Jon Sch?pzinsky > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/705e3f42/attachment-0001.html From msc at freeswitch.org Thu Jan 10 20:45:54 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Jan 2013 09:45:54 -0800 Subject: [Freeswitch-users] Early media without bridge In-Reply-To: <50EEA4D0.4020208@gmail.com> References: <50EEA4D0.4020208@gmail.com> Message-ID: What constitutes a failure? What I mean by that is, are *all* calls that have early media a "failed" call? Or just certain calls? What about ringing in early media? You could use monitor_early_media_fail and attempt to analyze the progress tones, i.e. if you hear ringing then it's a good call but otherwise it's a "failure." However, I'm not aware of any way to have the early media flow to the calling party with having the bridge/originate be "successful." The only way to know for sure is to check for a hangup cause. Could you explain more about what you are trying to accomplish? Perhaps there is another way of dealing with the early media conundrum. -MC On Thu, Jan 10, 2013 at 3:24 AM, Tamas.Cseke wrote: > Hello, > > We would like to hear early media without CHANNEL_BRIDGE event > These are failed calls and callers would like to hear the message that > the provider plays > Because the caller thinks the call is answered if originate returns. > > as far as I understand: > -early media makes the originate return > -if we ignore early media the bridge won't return, but we don't hear it > > we would like both of them, is it possible somehow? > > I 'm not sure I fully understand all of the ignore_early_media options > but I haven't find solution for this, > Could you please advise me one, if there is any? > > I'm thinking about we maybe need a new ignore_early_media option > like "consume" but sending the media to the caller instead of dropping it > If there isn't already a solution I also would appreciate if you let me > know your opinion about this idea > > Thanks advance, > Tamas > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/c03497a7/attachment.html From jmesquita at freeswitch.org Thu Jan 10 20:46:02 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 10 Jan 2013 14:46:02 -0300 Subject: [Freeswitch-users] Loopback Endpoint In-Reply-To: References: <111509FCE9EBF74BAB2CB38AAC521D9B23DCDDF44B@helle.dk.hq.firstcom.dk> Message-ID: Jon, the loopback will most certainly be evil if you are processing CDRs. If you want to keep it simple, avoid using it at all costs. It should always be a last resort. Jo?o Mesquita FreeSWITCH? Solutions On Thu, Jan 10, 2013 at 3:37 PM, Michael Collins wrote: > The devs have done their best to make loopback not be evil but in some > respects it must be, because it is what is it. Some folks use it all the > time without any issues. Just manage your expectations. :) > > That being said, you mentioned you need to dial a "dialplan" for each user > and that you need to do so in parallel. Without more information I can't > offer a better solution, although when I hear things like "dial in > parallel" I immediately think "enterprise originate." More information is > found on the wikior in the FS > Cookbook . > > -MC > > > On Thu, Jan 10, 2013 at 5:36 AM, Jon Sch?pzinsky wrote: > >> Hello List, >> >> I can see that the "this will destroy the world and this may kill your >> pets" warning has been removed from the documentation for the Loopback >> endpoint. >> >> Is this an indication that it has become safer to use? >> >> I have a specific problem where I essentially have to dial a dialplan for >> each user I am trying to reach, in parallel. >> Is it correctly understood that loopback would be the best/only way of >> implementing this, or is there another way? >> >> >> Kind Regards >> >> Jon Sch?pzinsky >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/487f385e/attachment.html From msc at freeswitch.org Thu Jan 10 20:47:18 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Jan 2013 09:47:18 -0800 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: Message-ID: Can you pastebin a console debug log w/ sip trace of a call failing to negotiate t.38 correctly? That might help us figure out where to look. -MC On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: > Just installed FusionPBX using the script built for a Centos 6 install. > Everything works great in regards to calls, but my main goal of this > project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have > tried many things in media bypass, routing to a fax extension, using an > ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me > in the right direction as to where I should begin? I am guessing it isn't > supposed to work right out of the box. I am using Flowroute as my provider > right now and the have confirmed that they utilize T38 in both directions. > Thanks! > > > Robert > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/07842d5b/attachment.html From jbaclor at ezuce.com Thu Jan 10 21:01:19 2013 From: jbaclor at ezuce.com (Joegen Baclor) Date: Fri, 11 Jan 2013 02:01:19 +0800 Subject: [Freeswitch-users] SIP packet exceeds MTU? In-Reply-To: <50EDA0A4.1010101@communicatefreely.net> References: <50EDA0A4.1010101@communicatefreely.net> Message-ID: <50EF01EF.8030107@ezuce.com> You don't have to agree to switch. The specs says you should 18.1.1 Sending Requests If a request is within 200 bytes of the path MTU, or if it is larger than 1300 bytes and the path MTU is unknown, the request MUST be sent using an RFC 2914 [43] congestion controlled transport protocol, such as TCP. On 01/10/2013 12:53 AM, Tim St. Pierre wrote: > Yes, I suppose I CAN switch to TCP, but I don't WANT to switch to TCP. > > My main gripe with TCP is that it holds a socket open for every registered endpoint. If I > have to restart a profile, I have to leave it down for several minutes until all the > sockets close before I can start it up again. > > I don't suppose there is a way to disable the and tags in the body? > > Thanks! > > > > Ken Rice wrote: >> Switch to TCP... This is why the RFCs tell you to support both UDP and TCP >> for SIP... >> >> >> On 1/9/13 9:36 AM, "Tim St. Pierre" wrote: >> >>> Hello, >>> >>> I'm having a problem with BLF in some cases, and I found that the length of >>> the username >>> was a factor in whether the lamps worked or not. >>> >>> When I did some packet captures, I found that problem packets were 1500 bytes >>> or so. >>> Since the endpoint is on a DSL line that has an MTU of 1460, the packet >>> doesn't fit. >>> >>> I'm already using compressed headers. Is there a way to reduce the packet >>> size more, or >>> is there another solution? I will include an example below: Note that my IP >>> address and >>> domain substutions may change the packet size a little, but the point is the >>> same. >>> >>> Thanks! >>> >>> NOTIFY sip:XXXX at YYY.YYY.YYY.YYY:5060;transport=udp SIP/2.0 >>> v:SIP/2.0/UDP zzz.zzz.zzz.zzz:5070;rport;branch=z9hG4bKD5QUr5m2Fm3Dp >>> Route:;transport=udp >>> Max-Forwards:70 >>> f:;tag=TrmHE4FBIrB4 >>> t:"Tim St. Pierre";tag=ecc2498bfd >>> i:3decd12decccafe1 >>> CSeq:448376002 NOTIFY >>> m:>> ip:yyy.yyy.yyy.yyy:61906%3Btransport%3Dudp> >>> User-Agent:Communicate Freely 2.0 >>> Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY, >>> PUBLISH,SUBSCRIBE >>> k:timer,precondition,path,replaces >>> o:dialog >>> u:talk,hold,conference,presence,dialog,line-seize,call-info,sla,include-sessio >>> n-description,presence.winfo,message-summary,refer >>> Subscription-State:active;expires=1137 >>> c:application/dialog-info+xml >>> l:639 >>> >>> >>> >> state="partial" >>> entity="sip:accountcode-701 at ourdomain.com"> >>> >>> confirmed >>> >>> >> display="park">sip:accountcode-701 at ourdomain.com;proto=park >>> >>> >>> >>> >>> >>> sip:accountcode-701 >>> >> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/c7210f7c/attachment-0001.html From lynn.nielson at greenseedtechnologies.com Thu Jan 10 21:14:07 2013 From: lynn.nielson at greenseedtechnologies.com (Lynn Nielson) Date: Thu, 10 Jan 2013 11:14:07 -0700 Subject: [Freeswitch-users] Use of Sangoma's Lyra AMD software Message-ID: We are looking for a good Advance Voicemail Detection solution and was wondering if anyone has integrated the Sangoma Lyra AMD solution into freeswitch? In talking to their sales agent, they indicated that it is only ported currently to Asterisk but that it would not be difficult to modify the sip headers to work with freeswitch. Thanks, -- Lynn Nielson Green Seed Technologies -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/4f2aa792/attachment.html From craig.robert at hotmail.com Thu Jan 10 23:43:09 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Thu, 10 Jan 2013 13:43:09 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: , Message-ID: Thanks for the quick response. Below is the capture from the log viewer. In this test, I am using a remote fax machine (line from Cox with physical machine) faxing into the "email server" where I created an extension for fax-to-email. Calling number is 1479-246-0547. 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 sofia/external/+14792460547 at flowroute.com Standard DESTROY 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/+14792460547 at flowroute.com SOFIA DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> DOWN 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 (sofia/external/+14792460547 at flowroute.com) Ended 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external entities 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> CS_DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING going to sleep 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: NORMAL_CLEARING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> CS_REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup Command with no Session system(/usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 caller_id_name='' caller_id_number=): 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: NORMAL_CLEARING 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to sofia/external/+14792460547 at flowroute.com 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel sofia/external/+14792460547 at flowroute.com hanging up, cause: NORMAL_CLEARING 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_HANGUP 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 sofia/external/+14792460547 at flowroute.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal sofia/external/+14792460547 at flowroute.com [KILL] 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP EXECUTE sofia/external/+14792460547 at flowroute.com hangup() 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 14400 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station id: ATW-4792460547 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (48) Disconnected after permitted retries. 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer spec [0] must be between 20 and 10000 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxRateManagement:transferredTCF m=image 31632 udptl t38 t=0 0 c=IN IP4 66.42.121.5 s=- o=- 275426305 1357849876 IN IP4 66.42.121.5 v=0 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completing][200] 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+14792460547 at flowroute.com Update Callee ID to "+14792460547" <+14792460547> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [calling][0] 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec activation Success L16 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 sofia/external/+14792460547 at flowroute.com Push codec L16:70 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec activation Success L16 20000 EXECUTE sofia/external/+14792460547 at flowroute.com rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done playing file silence_stream://2000 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms EXECUTE sofia/external/+14792460547 at flowroute.com playback(silence_stream://2000) 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [last_fax]=[+14792460547-2013-01-10-13-39-26] EXECUTE sofia/external/+14792460547 at flowroute.com set(last_fax=+14792460547-2013-01-10-13-39-26) 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38_request=true) 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38]=[true] 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completed][200] EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38=true) 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel [sofia/external/+14792460547 at flowroute.com] has been answered 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=sendrecv a=ptime:20 a=silenceSupp:off - - - - a=fmtp:101 0-16 a=rtpmap:101 telephone-event/8000 a=rtpmap:0 PCMU/8000 m=audio 27588 RTP/AVP 0 101 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP sofia/external/+14792460547 at flowroute.com: 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success [72.201.112.131]:[27588] EXECUTE sofia/external/+14792460547 at flowroute.com answer() 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=${last_fax} messages='result: ${fax_result_text} sender:${fax_remote_station_id} pages:${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] EXECUTE sofia/external/+14792460547 at flowroute.com set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} messages='result: \${fax_result_text} sender:\${fax_remote_station_id} pages:\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/+14792460547/3150) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [call_direction]=[local] EXECUTE sofia/external/+14792460547 at flowroute.com set(call_direction=local) 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() Dialplan: sofia/external/+14792460547 at flowroute.com Action rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) Dialplan: sofia/external/+14792460547 at flowroute.com Action playback(silence_stream://2000) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38_request=true) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38=true) Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() Dialplan: sofia/external/+14792460547 at flowroute.com Action set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} pages:\\\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\\\${caller_id_name}' caller_id_number=\\\${caller_id_number}) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test Fax] destination_number(3150) =~ /^3150$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Test Fax] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain_user] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->send_to_voicemail] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->extension-intercom] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->intercept-ext] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_return] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_privacy] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [global] Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->global] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [redial] destination_number(3150) =~ /^(redial|\*870)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->redial] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->group-intercept] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action set(call_direction=local) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call-direction] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Flowroute.11d] continue=false 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->3150 in context default 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer sofia/external/+14792460547 at flowroute.com to XML[3150 at default] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML default) 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] EXECUTE sofia/external/+14792460547 at flowroute.com export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [outside_call]=[true] EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 XML default) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] context(public) =~ /public/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->Test_Number] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->call_debug] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(outside_call=true) Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [outside_call] Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->outside_call] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->unloop] continue=false 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->15203039810 in context public 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+14792460547 at flowroute.com) Callstate Change DOWN -> RINGING 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+14792460547 at flowroute.com SOFIA INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf send/recv payload to 101 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 sofia/external/+14792460547 at flowroute.com Original read codec set to PCMU:0 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples 64000 bits 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] a=ptime:20 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=rtpmap:18 G729/8000 m=audio 31632 RTP/AVP 0 18 101 t=0 0 c=IN IP4 66.42.121.5 s=- o=- 275426305 1357849875 IN IP4 66.42.121.5 v=0 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [received][100] 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 (sofia/external/+14792460547 at flowroute.com) State NEW 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel sofia/external/+14792460547 at flowroute.com [d276f012-5b65-11e2-beae-8564ff827a25] Robert Date: Thu, 10 Jan 2013 09:47:18 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Can you pastebin a console debug log w/ sip trace of a call failing to negotiate t.38 correctly? That might help us figure out where to look. -MC On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: Just installed FusionPBX using the script built for a Centos 6 install. Everything works great in regards to calls, but my main goal of this project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have tried many things in media bypass, routing to a fax extension, using an ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me in the right direction as to where I should begin? I am guessing it isn't supposed to work right out of the box. I am using Flowroute as my provider right now and the have confirmed that they utilize T38 in both directions. Thanks! Robert _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/2a4a00c8/attachment-0001.html From jos at firstcom.dk Fri Jan 11 01:21:21 2013 From: jos at firstcom.dk (=?iso-8859-1?Q?Jon_Sch=F8pzinsky?=) Date: Thu, 10 Jan 2013 23:21:21 +0100 Subject: [Freeswitch-users] Loopback Endpoint In-Reply-To: References: <111509FCE9EBF74BAB2CB38AAC521D9B23DCDDF44B@helle.dk.hq.firstcom.dk> Message-ID: <111509FCE9EBF74BAB2CB38AAC521D9B23DCDDF4A4@helle.dk.hq.firstcom.dk> Hello Michael, I have taken a look at enteprise origination, and that seems like a solution. I was overthinking the solution, as I only need to originate to different devices in sequence, for each user, and enterprise origination works fine with that. Thank you :) Kind Regards Jon Sch?pzinsky Fra: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Michael Collins Sendt: 10. januar 2013 18:38 Til: FreeSWITCH Users Help Emne: Re: [Freeswitch-users] Loopback Endpoint The devs have done their best to make loopback not be evil but in some respects it must be, because it is what is it. Some folks use it all the time without any issues. Just manage your expectations. :) That being said, you mentioned you need to dial a "dialplan" for each user and that you need to do so in parallel. Without more information I can't offer a better solution, although when I hear things like "dial in parallel" I immediately think "enterprise originate." More information is found on the wiki or in the FS Cookbook. -MC On Thu, Jan 10, 2013 at 5:36 AM, Jon Sch?pzinsky > wrote: Hello List, I can see that the "this will destroy the world and this may kill your pets" warning has been removed from the documentation for the Loopback endpoint. Is this an indication that it has become safer to use? I have a specific problem where I essentially have to dial a dialplan for each user I am trying to reach, in parallel. Is it correctly understood that loopback would be the best/only way of implementing this, or is there another way? Kind Regards Jon Sch?pzinsky _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -- This message has been scanned for viruses and dangerous content, and is believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/7cbe331d/attachment.html From dujinfang at gmail.com Fri Jan 11 03:24:35 2013 From: dujinfang at gmail.com (Seven Du) Date: Fri, 11 Jan 2013 08:24:35 +0800 Subject: [Freeswitch-users] Early media without bridge In-Reply-To: References: <50EEA4D0.4020208@gmail.com> Message-ID: <531184DB5C7B47D9BFF281DB2FA21CEF@gmail.com> I think I understand Tamas as I have similar question. I think the CHANNEL_BRIDGE event fires when you get media from the b-leg, and 2 CHANNEL_ANSWER event fired on both channels when the b-leg answered. ignore_early_media on this case can prevent CHANNEL_BRIDGE from fire until the b-leg really answers the call. however, In some cased we'd like the a leg hear the early media, but get the CHANNEL_BRIDGE when b-leg is really answer the call. I thought it maybe impossible for now for the a leg to hear early media on a unbridged channel. Or maybe there's an option enable this I can't remember. FS considers a call/bridge is "success" when get early media, e.g. bridge user/b1|user/b2 when user/b1 sends early media it never has a chance to call user/b2. If user/b plays "the line is busy, please call later ?. ", and the a leg can hear that will be helpful, and for a patient a-leg, it will eventually can b2 when b1 timeout. the bridge user/b1,user/b2 case may be a little harder, what if both sends back early media, do we choose one to pass to the a leg, or mix them? or make a option to select it? so I think Tamas' question is - is it possible to work exactly as the same as the ignore_early_media scenario, but let the a leg hear early media. Thanks. -- Seven Du Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, January 11, 2013 at 1:45 AM, Michael Collins wrote: > What constitutes a failure? What I mean by that is, are *all* calls that have early media a "failed" call? Or just certain calls? What about ringing in early media? > > You could use monitor_early_media_fail and attempt to analyze the progress tones, i.e. if you hear ringing then it's a good call but otherwise it's a "failure." However, I'm not aware of any way to have the early media flow to the calling party with having the bridge/originate be "successful." The only way to know for sure is to check for a hangup cause. > > Could you explain more about what you are trying to accomplish? Perhaps there is another way of dealing with the early media conundrum. > > -MC > > On Thu, Jan 10, 2013 at 3:24 AM, Tamas.Cseke wrote: > > Hello, > > > > We would like to hear early media without CHANNEL_BRIDGE event > > These are failed calls and callers would like to hear the message that > > the provider plays > > Because the caller thinks the call is answered if originate returns. > > > > as far as I understand: > > -early media makes the originate return > > -if we ignore early media the bridge won't return, but we don't hear it > > > > we would like both of them, is it possible somehow? > > > > I 'm not sure I fully understand all of the ignore_early_media options > > but I haven't find solution for this, > > Could you please advise me one, if there is any? > > > > I'm thinking about we maybe need a new ignore_early_media option > > like "consume" but sending the media to the caller instead of dropping it > > If there isn't already a solution I also would appreciate if you let me > > know your opinion about this idea > > > > Thanks advance, > > Tamas > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/2a4a58d6/attachment.html From itsusama at gmail.com Fri Jan 11 04:26:07 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Fri, 11 Jan 2013 06:26:07 +0500 Subject: [Freeswitch-users] mod_conference auto-record issue Message-ID: Hi, Following in my conference profile: I'm having issues with auto-record, I keep getting these messages on the console: 2013-01-11 06:19:27.082662 [ERR] mod_conference.c:3884 Error Opening File [/root/home/test.wav] 2013-01-11 06:19:27.082662 [ERR] switch_core_timer.c:117 Timer is not properly configured. And if I keep it running for a while, FS segfaults. I was trying recording to a stream initially, but then to narrow down the issue I tried recording to a file and got the same error. Please advise. Regards. -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/2dd72df1/attachment-0001.html From fs-list at communicatefreely.net Fri Jan 11 04:37:48 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 10 Jan 2013 20:37:48 -0500 Subject: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure Message-ID: <50EF6CEC.1090503@communicatefreely.net> Okay, so I have a bit more detail, and my first post hasn't posted yet. FreeSWITCH Version 1.3.13b+git~20121230T190855Z~8859eb0b23 (git 8859eb0 2012-12-30 19:08:55Z FreeBSD 8.3-RELEASE I start getting thread_failures several times per second. It was so many entries that the logs started turning over at about 10 minute intervals. A few minutes later, I get the above quotes from Star Wars. After an hour and a half of no phone service, someone finds my pager number and I take a look. Our alarms didn't go off, because Freeswitch would still respond to sockets, and it was still acknowledging SIP messages to a certain degree. fs_cli times out. Freeswitch is using 96% CPU and around 900MB of RAM. Earlier today, it was 500MB, but I have seen it higher than that before. I have never seen it use more than 10% CPU on our little dual core Xeon machine. I probably should get more cores, but it has never gone past 10% CPU usage. So I kill -9 Freeswitch, wait a few seconds, and start it up again. Everything comes back after the phones re-register, but I would really like to know why, and how to prevent it from happening again. This isn't an especially high call volume night, and I looked through the logs - it started just after a new session was created for an outgoing call. Nothing unusual about it. For a while after, I could still see SIP registrations in the logs, but there weren't any SIP registrations in the database. The process to delete the expired ones seems to have been working, but not the process to update them. This is essentially a forensic investigation at this point, but I wouldn't mind some suggestions as to what I should watch out for. -Tim From fs-list at communicatefreely.net Fri Jan 11 04:44:49 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 10 Jan 2013 20:44:49 -0500 Subject: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure In-Reply-To: <50EF6CEC.1090503@communicatefreely.net> References: <50EF6CEC.1090503@communicatefreely.net> Message-ID: <50EF6E91.5090408@communicatefreely.net> Thought this might be helpful - switch_core_session.c:1655 Thread Failure! That's the actual error with the line of code. Tim St. Pierre wrote: > Okay, so I have a bit more detail, and my first post hasn't posted yet. > > FreeSWITCH Version 1.3.13b+git~20121230T190855Z~8859eb0b23 (git 8859eb0 2012-12-30 19:08:55Z > FreeBSD 8.3-RELEASE > > I start getting thread_failures several times per second. It was so many entries that the > logs started turning over at about 10 minute intervals. > > A few minutes later, I get the above quotes from Star Wars. > > After an hour and a half of no phone service, someone finds my pager number and I take a look. > > Our alarms didn't go off, because Freeswitch would still respond to sockets, and it was > still acknowledging SIP messages to a certain degree. > > fs_cli times out. > Freeswitch is using 96% CPU and around 900MB of RAM. Earlier today, it was 500MB, but I > have seen it higher than that before. I have never seen it use more than 10% CPU on our > little dual core Xeon machine. I probably should get more cores, but it has never gone > past 10% CPU usage. > > So I kill -9 Freeswitch, wait a few seconds, and start it up again. Everything comes back > after the phones re-register, but I would really like to know why, and how to prevent it > from happening again. > > This isn't an especially high call volume night, and I looked through the logs - it > started just after a new session was created for an outgoing call. Nothing unusual about > it. For a while after, I could still see SIP registrations in the logs, but there weren't > any SIP registrations in the database. The process to delete the expired ones seems to > have been working, but not the process to update them. > > This is essentially a forensic investigation at this point, but I wouldn't mind some > suggestions as to what I should watch out for. > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Fri Jan 11 04:46:12 2013 From: dujinfang at gmail.com (Seven Du) Date: Fri, 11 Jan 2013 09:46:12 +0800 Subject: [Freeswitch-users] mod_conference auto-record issue In-Reply-To: References: Message-ID: <87D3456F11834337803D77CC0DEE5E35@gmail.com> maybe check if you have the /root/home dir and is writable by the user running FreeSWITCH? -- Seven Du Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, January 11, 2013 at 9:26 AM, Usama Zaidi wrote: > Hi, > > Following in my conference profile: > > > > > > > > > > > I'm having issues with auto-record, I keep getting these messages on the console: > > 2013-01-11 06:19:27.082662 [ERR] mod_conference.c:3884 Error Opening File [/root/home/test.wav] > 2013-01-11 06:19:27.082662 [ERR] switch_core_timer.c:117 Timer is not properly configured. > > And if I keep it running for a while, FS segfaults. I was trying recording to a stream initially, but then to narrow down the issue I tried recording to a file and got the same error. Please advise. > > Regards. > > -- > I'd love to change the world, but they wont gimme the source code to it > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/ef94b37c/attachment.html From fs-list at communicatefreely.net Fri Jan 11 04:54:03 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 10 Jan 2013 20:54:03 -0500 Subject: [Freeswitch-users] PHP ESL timeout? Message-ID: <50EF70BB.4060800@communicatefreely.net> Hello, I currently have a testing system that opens an ESL connection to freeswitch to launch test calls, and also to make sure the switch is actually running as it should be. If the switch is up, it works. If the switch is totally down, it works. If the switch is partially up, it doesn't work. If the switch is up, but not really responsive, I am able to create a connection, but when I issue an API call, it sits there forever and doesn't return anything. If the API execute timed out and the script continued to execute, it would check the response, see that it is not what it should be, and alert the right people. Anyone know this library that can help? -Tim From peter.olsson at visionutveckling.se Fri Jan 11 09:40:29 2013 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 11 Jan 2013 06:40:29 +0000 Subject: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure In-Reply-To: <50EF6CEC.1090503@communicatefreely.net> References: <50EF6CEC.1090503@communicatefreely.net> Message-ID: <7A16DFEE-FC2F-4795-8FD1-EC5C9174D660@visionutveckling.se> Please report to JIRA. /Peter 11 jan 2013 kl. 02:43 skrev "Tim St. Pierre" : > Okay, so I have a bit more detail, and my first post hasn't posted yet. > > FreeSWITCH Version 1.3.13b+git~20121230T190855Z~8859eb0b23 (git 8859eb0 2012-12-30 19:08:55Z > FreeBSD 8.3-RELEASE > > I start getting thread_failures several times per second. It was so many entries that the > logs started turning over at about 10 minute intervals. > > A few minutes later, I get the above quotes from Star Wars. > > After an hour and a half of no phone service, someone finds my pager number and I take a look. > > Our alarms didn't go off, because Freeswitch would still respond to sockets, and it was > still acknowledging SIP messages to a certain degree. > > fs_cli times out. > Freeswitch is using 96% CPU and around 900MB of RAM. Earlier today, it was 500MB, but I > have seen it higher than that before. I have never seen it use more than 10% CPU on our > little dual core Xeon machine. I probably should get more cores, but it has never gone > past 10% CPU usage. > > So I kill -9 Freeswitch, wait a few seconds, and start it up again. Everything comes back > after the phones re-register, but I would really like to know why, and how to prevent it > from happening again. > > This isn't an especially high call volume night, and I looked through the logs - it > started just after a new session was created for an outgoing call. Nothing unusual about > it. For a while after, I could still see SIP registrations in the logs, but there weren't > any SIP registrations in the database. The process to delete the expired ones seems to > have been working, but not the process to update them. > > This is essentially a forensic investigation at this point, but I wouldn't mind some > suggestions as to what I should watch out for. > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:50ef69f032762133211294! > From devel at omninet.eu Fri Jan 11 10:22:07 2013 From: devel at omninet.eu (Anestis Mavro) Date: Fri, 11 Jan 2013 09:22:07 +0200 Subject: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure In-Reply-To: <7A16DFEE-FC2F-4795-8FD1-EC5C9174D660@visionutveckling.se> References: <50EF6CEC.1090503@communicatefreely.net> <7A16DFEE-FC2F-4795-8FD1-EC5C9174D660@visionutveckling.se> Message-ID: <96FD6925CA2A44F7A0DB5EF8B7B11A41@omni1.local> Hello Peter, I faced a very similar (if not the same) situation a few weeks ago. I had exactly the same behavior with the stable branch, but with the CPU very very low. This is on CentOS (with a lot of cores and memory) I went back to version 89e163a100 (November 11th, 2012) and the problem was gone. A few days ago I updated to master git 40cbc5c0e5 (January 4th, 2013) and the problem doesn't exist anymore. For me, it looks like there was something limiting the number of responses from FS to the clients. (New configuration for ODBC was already done!) I would try to upgrade to latest git. Good luck -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Friday, January 11, 2013 8:40 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure Please report to JIRA. /Peter 11 jan 2013 kl. 02:43 skrev "Tim St. Pierre" : > Okay, so I have a bit more detail, and my first post hasn't posted yet. > > FreeSWITCH Version 1.3.13b+git~20121230T190855Z~8859eb0b23 (git 8859eb0 2012-12-30 19:08:55Z > FreeBSD 8.3-RELEASE > > I start getting thread_failures several times per second. It was so many entries that the > logs started turning over at about 10 minute intervals. > > A few minutes later, I get the above quotes from Star Wars. > > After an hour and a half of no phone service, someone finds my pager number and I take a look. > > Our alarms didn't go off, because Freeswitch would still respond to sockets, and it was > still acknowledging SIP messages to a certain degree. > > fs_cli times out. > Freeswitch is using 96% CPU and around 900MB of RAM. Earlier today, it was 500MB, but I > have seen it higher than that before. I have never seen it use more than 10% CPU on our > little dual core Xeon machine. I probably should get more cores, but it has never gone > past 10% CPU usage. > > So I kill -9 Freeswitch, wait a few seconds, and start it up again. Everything comes back > after the phones re-register, but I would really like to know why, and how to prevent it > from happening again. > > This isn't an especially high call volume night, and I looked through the logs - it > started just after a new session was created for an outgoing call. Nothing unusual about > it. For a while after, I could still see SIP registrations in the logs, but there weren't > any SIP registrations in the database. The process to delete the expired ones seems to > have been working, but not the process to update them. > > This is essentially a forensic investigation at this point, but I wouldn't mind some > suggestions as to what I should watch out for. > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:50ef69f032762133211294! > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From stephen at picardogroup.com Fri Jan 11 05:03:32 2013 From: stephen at picardogroup.com (stephen at picardogroup.com) Date: Thu, 10 Jan 2013 19:03:32 -0700 Subject: [Freeswitch-users] =?utf-8?q?fs=5Fcli?= Message-ID: <20130110190332.0e1bd4d5c5064b420440751b21b10e46.3634fe1fed.wbe@email13.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130110/9e0f1285/attachment.html From avi at avimarcus.net Fri Jan 11 11:57:24 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 11 Jan 2013 10:57:24 +0200 Subject: [Freeswitch-users] fs_cli In-Reply-To: <20130110190332.0e1bd4d5c5064b420440751b21b10e46.3634fe1fed.wbe@email13.secureserver.net> References: <20130110190332.0e1bd4d5c5064b420440751b21b10e46.3634fe1fed.wbe@email13.secureserver.net> Message-ID: Check the end of your freeswitch.log file (not necessarily helpful) and also find the core file and post a backtrace of the bug to jira: http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy -Avi On Fri, Jan 11, 2013 at 4:03 AM, wrote: > Trying to run Fs_cli console. Machine was running last night. I installed > mod_lcrwithout any problem. This afternoon Fs seems not to be running. Any > directionwould be helpful. > > root at New-Host-2 [Thu Jan 10 05:59 PM] ~ > <3>:fs_cli -d 7 > [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is > /root/.fs_c li_conf. > [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is > /etc/fs_cli .conf. > [DEBUG] fs_cli.c:1306 main() profile default does not exist using builtin > profil e > [DEBUG] fs_cli.c:1333 main() Using profile internal [127.0.0.1] > [ERROR] fs_cli.c:1358 main() Error Connecting [Socket Connection Error] > Usage: fs_cli [-H ] [-P ] [-p ] [-d ] [-x > command] [- t ] [profile] > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/f11f854e/attachment.html From itsusama at gmail.com Fri Jan 11 12:02:01 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Fri, 11 Jan 2013 14:02:01 +0500 Subject: [Freeswitch-users] mod_conference auto-record issue (Seven Du) Message-ID: Hi Seven, Its not a file permissions Issue, its running under 'root' and I'm getting the same error if I use stream source instead of a file, the error message changes to: [ERR] mod_conference.c:3884 Error Opening File #transcode{vcodec=mp4v, acodec=mpga,vb=800,ab=128,deinterlace}: rtp{mux=ts,dst=192.168.1.8,port=5004,sdp=sap://,name=\"TestStream\"}] [ERR] switch_core_timer.c:117 Timer is not properly configured. Someone had the same error previously, not sure how they fixed it though, http://lists.freeswitch.org/pipermail/freeswitch-users/2011-June/073522.html On Fri, Jan 11, 2013 at 12:22 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Re: mod_conference auto-record issue (Seven Du) -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/65bf68fd/attachment.html From itsusama at gmail.com Fri Jan 11 12:08:10 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Fri, 11 Jan 2013 14:08:10 +0500 Subject: [Freeswitch-users] mod_conference auto-record issue (Seven Du) Message-ID: Ok, so turns out it was to do with file permissions, there was no folder /root/home/ creating the folder fixed the issue, any help regarding using mod_vlc to stream out, I can't use mod_shout as I don't want to transcode. On Fri, Jan 11, 2013 at 12:22 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. LUKE: I'm hit, but not bad - thread failure (Tim St. Pierre) > 2. Re: LUKE: I'm hit, but not bad - thread failure (Tim St. Pierre) > 3. Re: mod_conference auto-record issue (Seven Du) > 4. PHP ESL timeout? (Tim St. Pierre) > 5. Re: LUKE: I'm hit, but not bad - thread failure (Peter Olsson) > 6. Re: LUKE: I'm hit, but not bad - thread failure (Anestis Mavro) > > > ---------- Forwarded message ---------- > From: "Tim St. Pierre" > To: FreeSWITCH Users Help > Cc: > Date: Thu, 10 Jan 2013 20:37:48 -0500 > Subject: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure > Okay, so I have a bit more detail, and my first post hasn't posted yet. > > FreeSWITCH Version 1.3.13b+git~20121230T190855Z~**8859eb0b23 (git 8859eb0 > 2012-12-30 19:08:55Z > FreeBSD 8.3-RELEASE > > I start getting thread_failures several times per second. It was so many > entries that the logs started turning over at about 10 minute intervals. > > A few minutes later, I get the above quotes from Star Wars. > > After an hour and a half of no phone service, someone finds my pager > number and I take a look. > > Our alarms didn't go off, because Freeswitch would still respond to > sockets, and it was still acknowledging SIP messages to a certain degree. > > fs_cli times out. > Freeswitch is using 96% CPU and around 900MB of RAM. Earlier today, it > was 500MB, but I have seen it higher than that before. I have never seen > it use more than 10% CPU on our little dual core Xeon machine. I probably > should get more cores, but it has never gone past 10% CPU usage. > > So I kill -9 Freeswitch, wait a few seconds, and start it up again. > Everything comes back after the phones re-register, but I would really > like to know why, and how to prevent it from happening again. > > This isn't an especially high call volume night, and I looked through the > logs - it started just after a new session was created for an outgoing > call. Nothing unusual about it. For a while after, I could still see SIP > registrations in the logs, but there weren't any SIP registrations in the > database. The process to delete the expired ones seems to have been > working, but not the process to update them. > > This is essentially a forensic investigation at this point, but I wouldn't > mind some suggestions as to what I should watch out for. > > -Tim > > > > > ---------- Forwarded message ---------- > From: "Tim St. Pierre" > To: FreeSWITCH Users Help > Cc: > Date: Thu, 10 Jan 2013 20:44:49 -0500 > Subject: Re: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure > Thought this might be helpful - > > switch_core_session.c:1655 Thread Failure! > > That's the actual error with the line of code. > > > > Tim St. Pierre wrote: > >> Okay, so I have a bit more detail, and my first post hasn't posted yet. >> >> FreeSWITCH Version 1.3.13b+git~20121230T190855Z~**8859eb0b23 (git >> 8859eb0 2012-12-30 19:08:55Z >> FreeBSD 8.3-RELEASE >> >> I start getting thread_failures several times per second. It was so many >> entries that the logs started turning over at about 10 minute intervals. >> >> A few minutes later, I get the above quotes from Star Wars. >> >> After an hour and a half of no phone service, someone finds my pager >> number and I take a look. >> >> Our alarms didn't go off, because Freeswitch would still respond to >> sockets, and it was still acknowledging SIP messages to a certain degree. >> >> fs_cli times out. >> Freeswitch is using 96% CPU and around 900MB of RAM. Earlier today, it >> was 500MB, but I have seen it higher than that before. I have never seen >> it use more than 10% CPU on our little dual core Xeon machine. I probably >> should get more cores, but it has never gone past 10% CPU usage. >> >> So I kill -9 Freeswitch, wait a few seconds, and start it up again. >> Everything comes back after the phones re-register, but I would really >> like to know why, and how to prevent it from happening again. >> >> This isn't an especially high call volume night, and I looked through the >> logs - it started just after a new session was created for an outgoing >> call. Nothing unusual about it. For a while after, I could still see SIP >> registrations in the logs, but there weren't any SIP registrations in the >> database. The process to delete the expired ones seems to have been >> working, but not the process to update them. >> >> This is essentially a forensic investigation at this point, but I >> wouldn't mind some suggestions as to what I should watch out for. >> >> -Tim >> >> ______________________________**______________________________** >> _____________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.**freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.**freeswitch.org >> http://lists.freeswitch.org/**mailman/listinfo/freeswitch-**users >> UNSUBSCRIBE:http://lists.**freeswitch.org/mailman/** >> options/freeswitch-users >> http://www.freeswitch.org >> > > > > > > ---------- Forwarded message ---------- > From: Seven Du > To: FreeSWITCH Users Help > Cc: > Date: Fri, 11 Jan 2013 09:46:12 +0800 > Subject: Re: [Freeswitch-users] mod_conference auto-record issue > maybe check if you have the /root/home dir and is writable by the user > running FreeSWITCH? > > -- > Seven Du > Sent with Sparrow > > On Friday, January 11, 2013 at 9:26 AM, Usama Zaidi wrote: > > Hi, > > Following in my conference profile: > > > > > > > > > > > I'm having issues with auto-record, I keep getting these messages on the > console: > > 2013-01-11 06:19:27.082662 [ERR] mod_conference.c:3884 Error Opening File > [/root/home/test.wav] > 2013-01-11 06:19:27.082662 [ERR] switch_core_timer.c:117 Timer is not > properly configured. > > And if I keep it running for a while, FS segfaults. I was trying recording > to a stream initially, but then to narrow down the issue I tried recording > to a file and got the same error. Please advise. > > Regards. > > -- > I'd love to change the world, but they wont gimme the source code to it > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: "Tim St. Pierre" > To: FreeSWITCH Users Help > Cc: > Date: Thu, 10 Jan 2013 20:54:03 -0500 > Subject: [Freeswitch-users] PHP ESL timeout? > Hello, > > I currently have a testing system that opens an ESL connection to > freeswitch to launch test calls, and also to make sure the switch is > actually running as it should be. > > If the switch is up, it works. If the switch is totally down, it works. > If the switch is partially up, it doesn't work. > > If the switch is up, but not really responsive, I am able to create a > connection, but when I issue an API call, it sits there forever and doesn't > return anything. If the API execute timed out and the script continued to > execute, it would check the response, see that it is not what it should be, > and alert the right people. > > Anyone know this library that can help? > > -Tim > > > > > ---------- Forwarded message ---------- > From: Peter Olsson > To: FreeSWITCH Users Help > Cc: > Date: Fri, 11 Jan 2013 06:40:29 +0000 > Subject: Re: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure > Please report to JIRA. > > /Peter > > 11 jan 2013 kl. 02:43 skrev "Tim St. Pierre" < > fs-list at communicatefreely.net>: > > > Okay, so I have a bit more detail, and my first post hasn't posted yet. > > > > FreeSWITCH Version 1.3.13b+git~20121230T190855Z~8859eb0b23 (git 8859eb0 > 2012-12-30 19:08:55Z > > FreeBSD 8.3-RELEASE > > > > I start getting thread_failures several times per second. It was so > many entries that the > > logs started turning over at about 10 minute intervals. > > > > A few minutes later, I get the above quotes from Star Wars. > > > > After an hour and a half of no phone service, someone finds my pager > number and I take a look. > > > > Our alarms didn't go off, because Freeswitch would still respond to > sockets, and it was > > still acknowledging SIP messages to a certain degree. > > > > fs_cli times out. > > Freeswitch is using 96% CPU and around 900MB of RAM. Earlier today, it > was 500MB, but I > > have seen it higher than that before. I have never seen it use more > than 10% CPU on our > > little dual core Xeon machine. I probably should get more cores, but it > has never gone > > past 10% CPU usage. > > > > So I kill -9 Freeswitch, wait a few seconds, and start it up again. > Everything comes back > > after the phones re-register, but I would really like to know why, and > how to prevent it > > from happening again. > > > > This isn't an especially high call volume night, and I looked through > the logs - it > > started just after a new session was created for an outgoing call. > Nothing unusual about > > it. For a while after, I could still see SIP registrations in the logs, > but there weren't > > any SIP registrations in the database. The process to delete the > expired ones seems to > > have been working, but not the process to update them. > > > > This is essentially a forensic investigation at this point, but I > wouldn't mind some > > suggestions as to what I should watch out for. > > > > -Tim > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > !DSPAM:50ef69f032762133211294! > > > > > > > ---------- Forwarded message ---------- > From: "Anestis Mavro" > To: "'FreeSWITCH Users Help'" > Cc: > Date: Fri, 11 Jan 2013 09:22:07 +0200 > Subject: Re: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure > > Hello Peter, > > I faced a very similar (if not the same) situation a few weeks ago. > I had exactly the same behavior with the stable branch, but with the CPU > very very low. This is on CentOS (with a lot of cores and memory) > > I went back to version 89e163a100 (November 11th, 2012) and the problem was > gone. > > A few days ago I updated to master git 40cbc5c0e5 (January 4th, 2013) and > the problem doesn't exist anymore. > > For me, it looks like there was something limiting the number of responses > from FS to the clients. (New configuration for ODBC was already done!) > > I would try to upgrade to latest git. > > Good luck > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter > Olsson > Sent: Friday, January 11, 2013 8:40 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure > > Please report to JIRA. > > /Peter > > 11 jan 2013 kl. 02:43 skrev "Tim St. Pierre" > : > > > Okay, so I have a bit more detail, and my first post hasn't posted yet. > > > > FreeSWITCH Version 1.3.13b+git~20121230T190855Z~8859eb0b23 (git 8859eb0 > 2012-12-30 19:08:55Z > > FreeBSD 8.3-RELEASE > > > > I start getting thread_failures several times per second. It was so many > entries that the > > logs started turning over at about 10 minute intervals. > > > > A few minutes later, I get the above quotes from Star Wars. > > > > After an hour and a half of no phone service, someone finds my pager > number and I take a look. > > > > Our alarms didn't go off, because Freeswitch would still respond to > sockets, and it was > > still acknowledging SIP messages to a certain degree. > > > > fs_cli times out. > > Freeswitch is using 96% CPU and around 900MB of RAM. Earlier today, it > was 500MB, but I > > have seen it higher than that before. I have never seen it use more than > 10% CPU on our > > little dual core Xeon machine. I probably should get more cores, but it > has never gone > > past 10% CPU usage. > > > > So I kill -9 Freeswitch, wait a few seconds, and start it up again. > Everything comes back > > after the phones re-register, but I would really like to know why, and > how > to prevent it > > from happening again. > > > > This isn't an especially high call volume night, and I looked through the > logs - it > > started just after a new session was created for an outgoing call. > Nothing unusual about > > it. For a while after, I could still see SIP registrations in the logs, > but there weren't > > any SIP registrations in the database. The process to delete the expired > ones seems to > > have been working, but not the process to update them. > > > > This is essentially a forensic investigation at this point, but I > wouldn't > mind some > > suggestions as to what I should watch out for. > > > > -Tim > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > !DSPAM:50ef69f032762133211294! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/83644dea/attachment-0001.html From steveayre at gmail.com Fri Jan 11 12:18:47 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 11 Jan 2013 09:18:47 +0000 Subject: [Freeswitch-users] fs_cli In-Reply-To: References: <20130110190332.0e1bd4d5c5064b420440751b21b10e46.3634fe1fed.wbe@email13.secureserver.net> Message-ID: The 'dmesg' command is also an easy way to check if there have been any segfault crashes (add -c to clear the history). Confirm FS is stopped using 'ps -C freeswitch'. A line with a number and 'freeswitch' means it is running, no output except for the header means it is not. Use '/etc/init.d/freeswitch start' to get FS running again. -Steve On 11 January 2013 08:57, Avi Marcus wrote: > Check the end of your freeswitch.log file (not necessarily helpful) and > also find the core file and post a backtrace of the bug to jira: > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy > > -Avi > > On Fri, Jan 11, 2013 at 4:03 AM, wrote: > >> Trying to run Fs_cli console. Machine was running last night. I installed >> mod_lcrwithout any problem. This afternoon Fs seems not to be running. Any >> directionwould be helpful. >> >> root at New-Host-2 [Thu Jan 10 05:59 PM] ~ >> <3>:fs_cli -d 7 >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /root/.fs_c li_conf. >> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is >> /etc/fs_cli .conf. >> [DEBUG] fs_cli.c:1306 main() profile default does not exist using builtin >> profil e >> [DEBUG] fs_cli.c:1333 main() Using profile internal [127.0.0.1] >> [ERROR] fs_cli.c:1358 main() Error Connecting [Socket Connection Error] >> Usage: fs_cli [-H ] [-P ] [-p ] [-d ] [-x >> command] [- t ] [profile] >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/177b1630/attachment.html From chrisbware at yahoo.it Fri Jan 11 12:30:52 2013 From: chrisbware at yahoo.it (Chris B. Ware) Date: Fri, 11 Jan 2013 09:30:52 +0000 (GMT) Subject: [Freeswitch-users] RTP traffic on LAn Message-ID: <1357896652.59537.YahooMailNeo@web171804.mail.ir2.yahoo.com> Hi all, I'm trying to find a way to let RTP traffic between two phones, registered to a public Freeswitch, on the same LAN, remain local. Usually phones are natted behind an ADSL router and using two RTP streams to speack each other (because, for example, are on different? rooms on the same office) consume a lot of bandwidth. Is there anyone who has found a solution to this problem? I was thinking on using SIP INFO for sending DTMF (in case customer have to use some phone features activated via phone), using private IP (no STUN on phones) and keeping FS out of RTP streaming. Any help will be appreciated. Chris ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/d2139203/attachment.html From cstomi.levlist at gmail.com Fri Jan 11 12:37:21 2013 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Fri, 11 Jan 2013 10:37:21 +0100 Subject: [Freeswitch-users] Early media without bridge In-Reply-To: <531184DB5C7B47D9BFF281DB2FA21CEF@gmail.com> References: <50EEA4D0.4020208@gmail.com> <531184DB5C7B47D9BFF281DB2FA21CEF@gmail.com> Message-ID: Hello, Yes, this was my problem and I found the solution bridge_early_media=true was what I needed http://wiki.freeswitch.org/wiki/Variable_bridge_early_media There are so many awsome early media option, I couldn't find it I hope it help you too! Thank you guys! Tamas On Fri, Jan 11, 2013 at 1:24 AM, Seven Du wrote: > I think I understand Tamas as I have similar question. > > I think the CHANNEL_BRIDGE event fires when you get media from the b-leg, > and 2 CHANNEL_ANSWER event fired on both channels when the b-leg answered. > > ignore_early_media on this case can prevent CHANNEL_BRIDGE from fire until > the b-leg really answers the call. > > however, In some cased we'd like the a leg hear the early media, but get > the CHANNEL_BRIDGE when b-leg is really answer the call. > > I thought it maybe impossible for now for the a leg to hear early media on > a unbridged channel. Or maybe there's an option enable this I can't > remember. > > FS considers a call/bridge is "success" when get early media, e.g. > > bridge user/b1|user/b2 when user/b1 sends early media it never has a > chance to call user/b2. If user/b plays "the line is busy, please call > later ?. ", and the a leg can hear that will be helpful, and for a patient > a-leg, it will eventually can b2 when b1 timeout. > > the bridge user/b1,user/b2 case may be a little harder, what if both sends > back early media, do we choose one to pass to the a leg, or mix them? or > make a option to select it? > > so I think Tamas' question is - is it possible to work exactly as the same > as the ignore_early_media scenario, but let the a leg hear early media. > > Thanks. > > -- > Seven Du > Sent with Sparrow > > On Friday, January 11, 2013 at 1:45 AM, Michael Collins wrote: > > What constitutes a failure? What I mean by that is, are *all* calls that > have early media a "failed" call? Or just certain calls? What about ringing > in early media? > > You could use monitor_early_media_fail and attempt to analyze the progress > tones, i.e. if you hear ringing then it's a good call but otherwise it's a > "failure." However, I'm not aware of any way to have the early media flow > to the calling party with having the bridge/originate be "successful." The > only way to know for sure is to check for a hangup cause. > > Could you explain more about what you are trying to accomplish? Perhaps > there is another way of dealing with the early media conundrum. > > -MC > > On Thu, Jan 10, 2013 at 3:24 AM, Tamas.Cseke wrote: > > Hello, > > We would like to hear early media without CHANNEL_BRIDGE event > These are failed calls and callers would like to hear the message that > the provider plays > Because the caller thinks the call is answered if originate returns. > > as far as I understand: > -early media makes the originate return > -if we ignore early media the bridge won't return, but we don't hear it > > we would like both of them, is it possible somehow? > > I 'm not sure I fully understand all of the ignore_early_media options > but I haven't find solution for this, > Could you please advise me one, if there is any? > > I'm thinking about we maybe need a new ignore_early_media option > like "consume" but sending the media to the caller instead of dropping it > If there isn't already a solution I also would appreciate if you let me > know your opinion about this idea > > Thanks advance, > Tamas > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/734a7e5b/attachment-0001.html From steveayre at gmail.com Fri Jan 11 13:07:32 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 11 Jan 2013 10:07:32 +0000 Subject: [Freeswitch-users] RTP traffic on LAn In-Reply-To: <1357896652.59537.YahooMailNeo@web171804.mail.ir2.yahoo.com> References: <1357896652.59537.YahooMailNeo@web171804.mail.ir2.yahoo.com> Message-ID: All you need to do is either set bypass_media=true, or bypass_media_after_bridge=true. The hard part is knowing when to do so. You'll need to compare the incoming and outgoing IP addresses to see if they're on the same network, and there's no way really to do so without knowing those networks in advance. The phones aren't able to tell you. If you set bypass_media and the phones can't route directly to each other then all that happens is nothing gets heard, it doesn't cause any error or fail the call. I was thinking on using SIP INFO for sending DTMF (in case customer have to > use some phone features activated via phone) bypass_media_after_bridge=true will let FS collect RTP including DTMF during a IVR menu, then bridge to an endpoint and only have media going directly between caller+callee. -Steve On 11 January 2013 09:30, Chris B. Ware wrote: > Hi all, > > I'm trying to find a way to let RTP traffic between two phones, registered > to a public Freeswitch, on the same LAN, remain local. > Usually phones are natted behind an ADSL router and using two RTP streams > to speack each other (because, for example, are on different > rooms on the same office) consume a lot of bandwidth. > > Is there anyone who has found a solution to this problem? > > I was thinking on using SIP INFO for sending DTMF (in case customer have > to use some phone features activated via phone), using > private IP (no STUN on phones) and keeping FS out of RTP streaming. > > Any help will be appreciated. > > Chris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/2a5d08c4/attachment.html From oej at edvina.net Fri Jan 11 13:08:40 2013 From: oej at edvina.net (Olle E. Johansson) Date: Fri, 11 Jan 2013 11:08:40 +0100 Subject: [Freeswitch-users] Happy New Year! In-Reply-To: <5D85F846-B9C3-41DC-9A7A-2E92F75793CD@freeswitch.org> References: <1357223798.55422.YahooMailNeo@web160805.mail.bf1.yahoo.com> <5D85F846-B9C3-41DC-9A7A-2E92F75793CD@freeswitch.org> Message-ID: <9992C614-05BA-405A-A719-7189A28C56D8@edvina.net> 4 jan 2013 kl. 15:47 skrev Brian West : > You are correct you will learn a lot by doing that task. Many of us, including Brian, started as "Bug marshals" and learned a lot in that process in the old Asterisk days. And see where we ended up, with piles of our own created bugs implemented in a massive amount of source code... :-) Yes, you learn a lot and contribute a lot working with the bug tracker. It's a good way to start contributing to a project. /O > > On Jan 3, 2013, at 11:04 AM, Gabriel Gunderson wrote: > >> On Thu, Jan 3, 2013 at 7:36 AM, Mick Stevens wrote: >>> Count me in, anything I can do to assist (in return for, hopefully, the >>> acquisition of knowledge)? >> >> I suspect that working in Jira is it's own reward in terms of >> knowledge acquisition. >> >> Dig in and let us know what you find. >> >> >> Best, >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > http://freeswitchcookbook.com > http://freeswitchbook.com > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/a8bd41fc/attachment.html From avi at avimarcus.net Fri Jan 11 13:19:27 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 11 Jan 2013 12:19:27 +0200 Subject: [Freeswitch-users] RTP traffic on LAn In-Reply-To: References: <1357896652.59537.YahooMailNeo@web171804.mail.ir2.yahoo.com> Message-ID: Seems there are two options... if the phones have a static location, you can set a user variable for location. If they match on both phones, then it's internal. Or, check the contact IP for both lines -- if both are the same IP, it's the same network. sofia_contact gets you the entire contact string... I don't see how to get the ip directly. You can do "sofia status profile internal reg 1000" but the you have to parse it. One caveat: if there's multiple registrations turned on, you can have a user's desk phone and a smartphone sip client on different IPs. So you'd have to calculate if it's safe to add the bypass on each leg separately. Does uuid_simplify help in this case? I see no documentation on how it worked and finding it in the source code wasn't immediately helpful. -Avi Marcus 1-718-989-9485 (USA) 1-866-202-5850 (USA & Canada Toll Free) 02-372-1570 (Israel) 020-3298-2875 (UK) On Fri, Jan 11, 2013 at 12:07 PM, Steven Ayre wrote: > All you need to do is either set bypass_media=true, > or bypass_media_after_bridge=true. > > The hard part is knowing when to do so. You'll need to compare the > incoming and outgoing IP addresses to see if they're on the same network, > and there's no way really to do so without knowing those networks in > advance. The phones aren't able to tell you. If you set bypass_media and > the phones can't route directly to each other then all that happens is > nothing gets heard, it doesn't cause any error or fail the call. > > I was thinking on using SIP INFO for sending DTMF (in case customer have >> to use some phone features activated via phone) > > > bypass_media_after_bridge=true will let FS collect RTP including DTMF > during a IVR menu, then bridge to an endpoint and only have media going > directly between caller+callee. > > -Steve > > > > > On 11 January 2013 09:30, Chris B. Ware wrote: > >> Hi all, >> >> I'm trying to find a way to let RTP traffic between two phones, >> registered to a public Freeswitch, on the same LAN, remain local. >> Usually phones are natted behind an ADSL router and using two RTP streams >> to speack each other (because, for example, are on different >> rooms on the same office) consume a lot of bandwidth. >> >> Is there anyone who has found a solution to this problem? >> >> I was thinking on using SIP INFO for sending DTMF (in case customer have >> to use some phone features activated via phone), using >> private IP (no STUN on phones) and keeping FS out of RTP streaming. >> >> Any help will be appreciated. >> >> Chris >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/61dd8135/attachment-0001.html From raimund.sacherer at logitravel.com Fri Jan 11 14:19:12 2013 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Fri, 11 Jan 2013 12:19:12 +0100 (CET) Subject: [Freeswitch-users] RTP traffic on LAn In-Reply-To: <9912268.2397.1357903115167.JavaMail.javamailuser@localhost> Message-ID: <14759821.2404.1357903150414.JavaMail.javamailuser@localhost> Hi, isn't there allready a protocol that does that? ICE? this is an excerpt from the SNOM wiki page (we have about 280 snom phones in action): --- snip --- ICE DESCRIPTION Choose whether or not you want to use ICE (Interactive Connectivity Establishment). ICE optimizes the media path. This would be the case, for example, when two phones in the same network are calling each other via a long media path through other, external networks. With ICE, the short media path in the same network would be chosen, which will presumably have better quality than the long one. Sometimes this feature will stop you from being able to make calls. When this occurs, switch it off. --- snip --- Link here: http://wiki.snom.com/Settings/user_ice Here seems to be more information, check "Probing Media Path": http://wiki.snom.com/Networking/NAT Best, Raimund ----- Original Message ----- From: "Chris B. Ware" To: freeswitch-users at lists.freeswitch.org Sent: Viernes, 11 de Enero 2013 10:30:52 Subject: [Freeswitch-users] RTP traffic on LAn Hi all, I'm trying to find a way to let RTP traffic between two phones, registered to a public Freeswitch, on the same LAN, remain local. Usually phones are natted behind an ADSL router and using two RTP streams to speack each other (because, for example, are on different rooms on the same office) consume a lot of bandwidth. Is there anyone who has found a solution to this problem? I was thinking on using SIP INFO for sending DTMF (in case customer have to use some phone features activated via phone), using private IP (no STUN on phones) and keeping FS out of RTP streaming. Any help will be appreciated. Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 S?guenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para m?vil Logitravel.com Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electr?nico por error, se ruega notif?quese inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formar?n parte de un fichero registrado como CLIENTES con n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a estos datos que deber? ejercer mediante escrito a la direcci?n anteriormente citada. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/b12217d0/attachment.html From raimund.sacherer at logitravel.com Fri Jan 11 14:55:25 2013 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Fri, 11 Jan 2013 12:55:25 +0100 (CET) Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: <29236127.2623.1357905322543.JavaMail.javamailuser@localhost> Message-ID: <4592719.2627.1357905323526.JavaMail.javamailuser@localhost> Does anyone have any experience with voip monitor? http://www.voipmonitor.org/ We will start evaluating it in our environment as we are switching more and more to SIP providers, therefore I would love to have better Monitoring. What voipmonitor says is that they are able to filter the calls in realtime with low overhead an if call quality drops, they can alert you, and they seem to have a rather nice web interface and they also are used by others to store the RTP's ... What seems to be especially nice is that their capture daemon (as they state it) adds very little overhead to a system, so depending on call quantity they say it's neglect-able. Would love to hear some comparisions between this and HOMER if some can provide it. Thank you, best Ray ----- Original Message ----- From: "Cal Leeming [Simplicity Media Ltd]" To: "FreeSWITCH Users Help" Sent: Domingo, 6 de Enero 2013 21:01:24 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension Could a core dev speculate on how much the expected bounty would be for this? Thanks Cal _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 S?guenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para m?vil Logitravel.com Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electr?nico por error, se ruega notif?quese inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formar?n parte de un fichero registrado como CLIENTES con n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a estos datos que deber? ejercer mediante escrito a la direcci?n anteriormente citada. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/5939ae34/attachment-0001.html From steveayre at gmail.com Fri Jan 11 15:34:15 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 11 Jan 2013 12:34:15 +0000 Subject: [Freeswitch-users] RTP traffic on LAn In-Reply-To: <14759821.2404.1357903150414.JavaMail.javamailuser@localhost> References: <9912268.2397.1357903115167.JavaMail.javamailuser@localhost> <14759821.2404.1357903150414.JavaMail.javamailuser@localhost> Message-ID: Unfortunately FS does not (as yet) support it. At least not AFAIK. It also would require that both endpoints support it too, and not all do. In any case since FS is a B2BUA not a SIP proxy the endpoints are negotiating with FS not the other endpoint - so you're still going to want to detect that you want to (try to) bypass_media. -Steve On 11 January 2013 11:19, Raimund Sacherer wrote: > Hi, > > isn't there allready a protocol that does that? ICE? > > this is an excerpt from the SNOM wiki page (we have about 280 snom phones > in action): > > --- snip --- > ICE DESCRIPTION Choose whether or not you want to use ICE (Interactive > Connectivity Establishment). ICE optimizes the media path. This would be > the case, for example, when two phones in the same network are calling each > other via a long media path through other, external networks. With ICE, the > short media path in the same network would be chosen, which will presumably > have better quality than the long one. Sometimes this feature will stop you > from being able to make calls. When this occurs, switch it off. > --- snip --- > > Link here: http://wiki.snom.com/Settings/user_ice > > Here seems to be more information, check "Probing Media Path": > http://wiki.snom.com/Networking/NAT > > Best, > > Raimund > > ------------------------------ > *From: *"Chris B. Ware" > *To: *freeswitch-users at lists.freeswitch.org > *Sent: *Viernes, 11 de Enero 2013 10:30:52 > *Subject: *[Freeswitch-users] RTP traffic on LAn > > > Hi all, > > I'm trying to find a way to let RTP traffic between two phones, registered > to a public Freeswitch, on the same LAN, remain local. > Usually phones are natted behind an ADSL router and using two RTP streams > to speack each other (because, for example, are on different > rooms on the same office) consume a lot of bandwidth. > > Is there anyone who has found a solution to this problem? > > I was thinking on using SIP INFO for sending DTMF (in case customer have > to use some phone features activated via phone), using > private IP (no STUN on phones) and keeping FS out of RTP streaming. > > Any help will be appreciated. > > Chris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > [image: Logitravel.com] Raimund Sacherer > *Sistemas*Agencia de Viajes Online > www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) > Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca > Tel 902 366 847 | Fax 971 213 495 S?guenos en:[image: Facebook de > Logitravel] [image: Twitter de > Logitravel] [image: Blog de Logitravel] [image: > Logitravel en Youtube] [image: > Logitravel en Foursquare] Descarga > nuestras *aplicaciones para m?vil* [image: Logitravel.com]Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene > informaci?n de car?cter confidencial exclusivamente dirigida a su > destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a > terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de > haber recibido este correo electr?nico por error, se ruega notif?quese > inmediatamente esta circunstancia mediante reenv?o a la direcci?n > electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus > datos forman o formar?n parte de un fichero registrado como CLIENTES con > n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de > Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio > Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. > Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a > estos datos que deber? ejercer mediante escrito a la direcci?n > anteriormente citada. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/fc3ce00e/attachment.html From raimund.sacherer at logitravel.com Fri Jan 11 15:47:45 2013 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Fri, 11 Jan 2013 13:47:45 +0100 (CET) Subject: [Freeswitch-users] RTP traffic on LAn In-Reply-To: <25553580.2801.1357908447940.JavaMail.javamailuser@localhost> Message-ID: <6050539.2806.1357908465711.JavaMail.javamailuser@localhost> Hi, I have not looked much into this, but I was under the impression that it's a client-side thing, as I read it the SDP contains various IP's for the media, the public (which the phone seeks out via STUN) and the private, the other phone tries to connect to each of those and the first answer it get's it uses for the media path, as a LAN should answer always faster than a WAN, it should create the mediapath directly between the two devices, therefore FS will not see the media and interacts only with the signaling. This is how I have interpreted ICE. Now, if it does not work this way I appologize, up to now I did not have any use for ICE as all our remote offices are small and they do normally not call each other and in our main offices our PBX's are (still) internal ... Best, Rai ----- Original Message ----- From: "Steven Ayre" To: "FreeSWITCH Users Help" Sent: Viernes, 11 de Enero 2013 13:34:15 Subject: Re: [Freeswitch-users] RTP traffic on LAn Unfortunately FS does not (as yet) support it. At least not AFAIK. It also would require that both endpoints support it too, and not all do. In any case since FS is a B2BUA not a SIP proxy the endpoints are negotiating with FS not the other endpoint - so you're still going to want to detect that you want to (try to) bypass_media. -Steve On 11 January 2013 11:19, Raimund Sacherer < raimund.sacherer at logitravel.com > wrote: Hi, isn't there allready a protocol that does that? ICE? this is an excerpt from the SNOM wiki page (we have about 280 snom phones in action): --- snip --- ICE DESCRIPTION Choose whether or not you want to use ICE (Interactive Connectivity Establishment). ICE optimizes the media path. This would be the case, for example, when two phones in the same network are calling each other via a long media path through other, external networks. With ICE, the short media path in the same network would be chosen, which will presumably have better quality than the long one. Sometimes this feature will stop you from being able to make calls. When this occurs, switch it off. --- snip --- Link here: http://wiki.snom.com/Settings/user_ice Here seems to be more information, check "Probing Media Path": http://wiki.snom.com/Networking/NAT Best, Raimund From: "Chris B. Ware" < chrisbware at yahoo.it > To: freeswitch-users at lists.freeswitch.org Sent: Viernes, 11 de Enero 2013 10:30:52 Subject: [Freeswitch-users] RTP traffic on LAn Hi all, I'm trying to find a way to let RTP traffic between two phones, registered to a public Freeswitch, on the same LAN, remain local. Usually phones are natted behind an ADSL router and using two RTP streams to speack each other (because, for example, are on different rooms on the same office) consume a lot of bandwidth. Is there anyone who has found a solution to this problem? I was thinking on using SIP INFO for sending DTMF (in case customer have to use some phone features activated via phone), using private IP (no STUN on phones) and keeping FS out of RTP streaming. Any help will be appreciated. Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 S?guenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para m?vil Logitravel.com Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electr?nico por error, se ruega notif?quese inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formar?n parte de un fichero registrado como CLIENTES con n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a estos datos que deber? ejercer mediante escrito a la direcci?n anteriormente citada. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 S?guenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para m?vil Logitravel.com Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electr?nico por error, se ruega notif?quese inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formar?n parte de un fichero registrado como CLIENTES con n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a estos datos que deber? ejercer mediante escrito a la direcci?n anteriormente citada. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/7e557490/attachment-0001.html From steveayre at gmail.com Fri Jan 11 16:13:11 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 11 Jan 2013 13:13:11 +0000 Subject: [Freeswitch-users] RTP traffic on LAn In-Reply-To: <6050539.2806.1357908465711.JavaMail.javamailuser@localhost> References: <25553580.2801.1357908447940.JavaMail.javamailuser@localhost> <6050539.2806.1357908465711.JavaMail.javamailuser@localhost> Message-ID: You need to understand that because FS is a B2BUA by default there are two separate media sessions endpoint1->fs and fs->endpoint2. They negotiate completely separately, and FS just copies media across during a bridge. That means endpoint2 never sees the SDP of endpoint1... unless you bypass_media. If you set bypass_media=true then they will see the actual SDP of the other endpoint (forwarded by FS). Reading RFC5245.it looks like that would be enough for it to work. But you'll still want to know *when* to bypass_media since it might not work in all situations especially when NAT is involved and when one endpoints doesn't support STUN/ICE. In a NAT situation you'll still be reliant on your router's SIP ALG being disabled - the endpoints are setting all their candidate addresses in the SDP and if the router were to modify them then that could break everything. In the case bypass_media_after_bridge=true might still allow RFC2833 DTMF for IVR menus hosted on FS before it leaves the media path. I'm not clear on how well ICE would work with the media renegotiation, but presumably it would work as well as the initial negotation. Chris, relying on SIP INFO isn't necessarily a good idea - not all endpoints would send it for a start, and it only ever made it as far as a IETF draft that was dropped... although plenty of people have implemented it it never became an actual standard. In particular there are timing issues with it if the packets arrive out of order due to network jitter, since it lacks the timestamping that RFC2833 provides. Far better to remain in the media path for IVR and then to bow out. You're going to be wanting to play prompts to the user during the IVR anyway. -Steve On 11 January 2013 12:47, Raimund Sacherer wrote: > Hi, > > I have not looked much into this, but I was under the impression that it's > a client-side thing, as I read it the SDP contains various IP's for the > media, the public (which the phone seeks out via STUN) and the private, the > other phone tries to connect to each of those and the first answer it get's > it uses for the media path, as a LAN should answer always faster than a > WAN, it should create the mediapath directly between the two devices, > therefore FS will not see the media and interacts only with the signaling. > > This is how I have interpreted ICE. > > Now, if it does not work this way I appologize, up to now I did not have > any use for ICE as all our remote offices are small and they do normally > not call each other and in our main offices our PBX's are (still) internal > ... > > Best, > Rai > > ------------------------------ > *From: *"Steven Ayre" > *To: *"FreeSWITCH Users Help" > *Sent: *Viernes, 11 de Enero 2013 13:34:15 > *Subject: *Re: [Freeswitch-users] RTP traffic on LAn > > > Unfortunately FS does not (as yet) support it. At least not AFAIK. > It also would require that both endpoints support it too, and not all do. > > In any case since FS is a B2BUA not a SIP proxy the endpoints are > negotiating with FS not the other endpoint - so you're still going to want > to detect that you want to (try to) bypass_media. > > -Steve > > > > > > On 11 January 2013 11:19, Raimund Sacherer < > raimund.sacherer at logitravel.com> wrote: > >> Hi, >> >> isn't there allready a protocol that does that? ICE? >> >> this is an excerpt from the SNOM wiki page (we have about 280 snom phones >> in action): >> >> --- snip --- >> ICE DESCRIPTION Choose whether or not you want to use ICE (Interactive >> Connectivity Establishment). ICE optimizes the media path. This would be >> the case, for example, when two phones in the same network are calling each >> other via a long media path through other, external networks. With ICE, the >> short media path in the same network would be chosen, which will presumably >> have better quality than the long one. Sometimes this feature will stop you >> from being able to make calls. When this occurs, switch it off. >> --- snip --- >> >> Link here: http://wiki.snom.com/Settings/user_ice >> >> Here seems to be more information, check "Probing Media Path": >> http://wiki.snom.com/Networking/NAT >> >> Best, >> >> Raimund >> >> ------------------------------ >> *From: *"Chris B. Ware" >> *To: *freeswitch-users at lists.freeswitch.org >> *Sent: *Viernes, 11 de Enero 2013 10:30:52 >> *Subject: *[Freeswitch-users] RTP traffic on LAn >> >> >> Hi all, >> >> I'm trying to find a way to let RTP traffic between two phones, >> registered to a public Freeswitch, on the same LAN, remain local. >> Usually phones are natted behind an ADSL router and using two RTP streams >> to speack each other (because, for example, are on different >> rooms on the same office) consume a lot of bandwidth. >> >> Is there anyone who has found a solution to this problem? >> >> I was thinking on using SIP INFO for sending DTMF (in case customer have >> to use some phone features activated via phone), using >> private IP (no STUN on phones) and keeping FS out of RTP streaming. >> >> Any help will be appreciated. >> >> Chris >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> [image: Logitravel.com] Raimund Sacherer >> *Sistemas*Agencia de Viajes Online >> www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) >> Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca >> Tel 902 366 847 | Fax 971 213 495 S?guenos en:[image: Facebook de >> Logitravel] [image: Twitter de >> Logitravel] [image: Blog de Logitravel] [image: >> Logitravel en Youtube] [image: >> Logitravel en Foursquare] Descarga >> nuestras *aplicaciones para m?vil* [image: Logitravel.com]Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene >> informaci?n de car?cter confidencial exclusivamente dirigida a su >> destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a >> terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de >> haber recibido este correo electr?nico por error, se ruega notif?quese >> inmediatamente esta circunstancia mediante reenv?o a la direcci?n >> electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus >> datos forman o formar?n parte de un fichero registrado como CLIENTES con >> n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de >> Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio >> Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. >> Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a >> estos datos que deber? ejercer mediante escrito a la direcci?n >> anteriormente citada. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > [image: Logitravel.com] Raimund Sacherer > *Sistemas*Agencia de Viajes Online > www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) > Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca > Tel 902 366 847 | Fax 971 213 495 S?guenos en:[image: Facebook de > Logitravel] [image: Twitter de > Logitravel] [image: Blog de Logitravel] [image: > Logitravel en Youtube] [image: > Logitravel en Foursquare] Descarga > nuestras *aplicaciones para m?vil* [image: Logitravel.com]Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene > informaci?n de car?cter confidencial exclusivamente dirigida a su > destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a > terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de > haber recibido este correo electr?nico por error, se ruega notif?quese > inmediatamente esta circunstancia mediante reenv?o a la direcci?n > electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus > datos forman o formar?n parte de un fichero registrado como CLIENTES con > n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de > Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio > Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. > Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a > estos datos que deber? ejercer mediante escrito a la direcci?n > anteriormente citada. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/8d031227/attachment-0001.html From chrisbware at yahoo.it Fri Jan 11 16:31:54 2013 From: chrisbware at yahoo.it (Chris B. Ware) Date: Fri, 11 Jan 2013 13:31:54 +0000 (GMT) Subject: [Freeswitch-users] RTP traffic on LAn Message-ID: <1357911114.62078.YahooMailNeo@web171806.mail.ir2.yahoo.com> Thank you guys for all your hints. ICE & TURN are quite new protocols invented to fix all issues that STUN leave not fixed, such as double NAT. They try to discover client network topology. I think best solution is marking each phone on the same site with a flag and using bypass media. Thanks again. Chris B. Ware -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/db384069/attachment.html From kbdfck at gmail.com Fri Jan 11 16:35:33 2013 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Fri, 11 Jan 2013 17:35:33 +0400 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: <4592719.2627.1357905323526.JavaMail.javamailuser@localhost> References: <29236127.2623.1357905322543.JavaMail.javamailuser@localhost> <4592719.2627.1357905323526.JavaMail.javamailuser@localhost> Message-ID: We use VoipMonitor open-source part, with homebrew custom WEB-UI. This is a really cool software, it continuously evolves and captures pretty well even large amounts of VoiP traffic. I highly recommend this software, since it doesn't only captures signaling traffic, but also can capture media/rtcp, automatically convert captured calls to wav files or store raw pcaps for further processing. it also can analyze MOS/Loss scores and many many other different parameters and store them in database per call. We use single instance to store data for investigation of calls with problems on more than 10 servers. 2013/1/11 Raimund Sacherer > Does anyone have any experience with voip monitor? > http://www.voipmonitor.org/ > > We will start evaluating it in our environment as we are switching more > and more to SIP providers, therefore I would love to have better > Monitoring. > > What voipmonitor says is that they are able to filter the calls in > realtime with low overhead an if call quality drops, they can alert you, > and they seem to have a rather nice web interface and they also are used by > others to store the RTP's ... > > What seems to be especially nice is that their capture daemon (as they > state it) adds very little overhead to a system, so depending on call > quantity they say it's neglect-able. > > Would love to hear some comparisions between this and HOMER if some can > provide it. > > Thank you, best > Ray > > ------------------------------ > *From: *"Cal Leeming [Simplicity Media Ltd]" < > cal.leeming at simplicitymedialtd.co.uk> > *To: *"FreeSWITCH Users Help" > *Sent: *Domingo, 6 de Enero 2013 21:01:24 > *Subject: *[Freeswitch-users] Bounty for - FS-4268 Sip trace with filter > for IP or extension > > > Could a core dev speculate on how much the expected bounty would be for > this? > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > [image: Logitravel.com] Raimund Sacherer > *Sistemas*Agencia de Viajes Online > www.logitravel.comEdificio Logitravel, Parcela 3B (Parc Bit) > Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca > Tel 902 366 847 | Fax 971 213 495S?guenos en:[image: Facebook de > Logitravel] [image: Twitter de > Logitravel] [image: Blog de Logitravel][image: > Logitravel en Youtube] [image: > Logitravel en Foursquare] Descarga > nuestras *aplicaciones para m?vil*[image: Logitravel.com]Este > correo electr?nico y, en su caso, cualquier fichero anexo, contiene > informaci?n de car?cter confidencial exclusivamente dirigida a su > destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a > terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de > haber recibido este correo electr?nico por error, se ruega notif?quese > inmediatamente esta circunstancia mediante reenv?o a la direcci?n > electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus > datos forman o formar?n parte de un fichero registrado como CLIENTES con > n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de > Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio > Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. > Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a > estos datos que deber? ejercer mediante escrito a la direcci?n > anteriormente citada. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Best regards, Dmitry Sytchev, IT Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/604aa4d4/attachment.html From andrew at cassidywebservices.co.uk Fri Jan 11 17:04:09 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 11 Jan 2013 14:04:09 +0000 Subject: [Freeswitch-users] RTP traffic on LAn In-Reply-To: <1357911114.62078.YahooMailNeo@web171806.mail.ir2.yahoo.com> References: <1357911114.62078.YahooMailNeo@web171806.mail.ir2.yahoo.com> Message-ID: I think this has been discussed once before, and perhaps the best/only way would be to use and on-bridge script to compare the addresses. However, when you call uuid_media and freeswitch sends the reinvite, I bet it using the resolved public IP address of the phones rather than the internal ones. If there's a way to do it, it'll be based on that. Running a script to send the reinvite pointing to the internal addresses and port numbers SDP addresses. Assuming the SDP hasn't been modified. Theres On 11 January 2013 13:31, Chris B. Ware wrote: > Thank you guys for all your hints. > > ICE & TURN are quite new protocols invented to fix all issues that STUN > leave not fixed, such as double NAT. > They try to discover client network topology. > > I think best solution is marking each phone on the same site with a flag > and using bypass media. > > Thanks again. > > Chris B. Ware > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/ce5508d3/attachment-0001.html From jmesquita at freeswitch.org Fri Jan 11 17:18:56 2013 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Fri, 11 Jan 2013 11:18:56 -0300 Subject: [Freeswitch-users] Happy New Year! In-Reply-To: <9992C614-05BA-405A-A719-7189A28C56D8@edvina.net> References: <1357223798.55422.YahooMailNeo@web160805.mail.bf1.yahoo.com> <5D85F846-B9C3-41DC-9A7A-2E92F75793CD@freeswitch.org> <9992C614-05BA-405A-A719-7189A28C56D8@edvina.net> Message-ID: I can say that I've been doing it on Jira and this is very true. You learn a lot without even asking the core devs to explain you things. It is an self teaching process but at the same time it does require some dedication. Above all it requires will. Sent from my iPhone On Jan 11, 2013, at 7:08 AM, "Olle E. Johansson" wrote: > > 4 jan 2013 kl. 15:47 skrev Brian West : > >> You are correct you will learn a lot by doing that task. > Many of us, including Brian, started as "Bug marshals" and learned a lot in that process in the old Asterisk days. > > And see where we ended up, with piles of our own created bugs implemented in a > massive amount of source code... :-) > > Yes, you learn a lot and contribute a lot working with the bug tracker. It's a good way to start contributing > to a project. > > /O >> >> On Jan 3, 2013, at 11:04 AM, Gabriel Gunderson wrote: >> >>> On Thu, Jan 3, 2013 at 7:36 AM, Mick Stevens wrote: >>>> Count me in, anything I can do to assist (in return for, hopefully, the >>>> acquisition of knowledge)? >>> >>> I suspect that working in Jira is it's own reward in terms of >>> knowledge acquisition. >>> >>> Dig in and let us know what you find. >>> >>> >>> Best, >>> Gabe >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX PO BOX 2531 >> Brookfield, WI 53008-2531 >> >> Twitter: @FreeSWITCH_Wire >> http://freeswitchcookbook.com >> http://freeswitchbook.com >> >> T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST >> iNUM: +883 5100 1286 0410 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/823c8cc4/attachment.html From kris at kriskinc.com Fri Jan 11 17:59:45 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 11 Jan 2013 09:59:45 -0500 Subject: [Freeswitch-users] RTP traffic on LAn In-Reply-To: References: <9912268.2397.1357903115167.JavaMail.javamailuser@localhost> <14759821.2404.1357903150414.JavaMail.javamailuser@localhost> Message-ID: This may change soon... If your endpoints support ICE FS can ignore it with bypass media enabled. On Fri, Jan 11, 2013 at 7:34 AM, Steven Ayre wrote: > > Unfortunately FS does not (as yet) support it. At least not AFAIK. > It also would require that both endpoints support it too, and not all do. > > In any case since FS is a B2BUA not a SIP proxy the endpoints are negotiating with FS not the other endpoint - so you're still going to want to detect that you want to (try to) bypass_media. > > -Steve > > > > > > On 11 January 2013 11:19, Raimund Sacherer wrote: >> >> Hi, >> >> isn't there allready a protocol that does that? ICE? >> >> this is an excerpt from the SNOM wiki page (we have about 280 snom phones in action): >> >> --- snip --- >> ICE DESCRIPTION Choose whether or not you want to use ICE (Interactive Connectivity Establishment). ICE optimizes the media path. This would be the case, for example, when two phones in the same network are calling each other via a long media path through other, external networks. With ICE, the short media path in the same network would be chosen, which will presumably have better quality than the long one. Sometimes this feature will stop you from being able to make calls. When this occurs, switch it off. >> --- snip --- >> >> Link here: http://wiki.snom.com/Settings/user_ice >> >> Here seems to be more information, check "Probing Media Path": http://wiki.snom.com/Networking/NAT >> >> Best, >> >> Raimund >> >> ________________________________ >> From: "Chris B. Ware" >> To: freeswitch-users at lists.freeswitch.org >> Sent: Viernes, 11 de Enero 2013 10:30:52 >> Subject: [Freeswitch-users] RTP traffic on LAn >> >> >> Hi all, >> >> I'm trying to find a way to let RTP traffic between two phones, registered to a public Freeswitch, on the same LAN, remain local. >> Usually phones are natted behind an ADSL router and using two RTP streams to speack each other (because, for example, are on different >> rooms on the same office) consume a lot of bandwidth. >> >> Is there anyone who has found a solution to this problem? >> >> I was thinking on using SIP INFO for sending DTMF (in case customer have to use some phone features activated via phone), using >> private IP (no STUN on phones) and keeping FS out of RTP streaming. >> >> Any help will be appreciated. >> >> Chris >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> Raimund Sacherer >> Sistemas >> Agencia de Viajes Online >> www.logitravel.com >> Edificio Logitravel, Parcela 3B (Parc Bit) >> Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca >> Tel 902 366 847 | Fax 971 213 495 >> S?guenos en: >> Descarga nuestras aplicaciones para m?vil >> Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electr?nico por error, se ruega notif?quese inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formar?n parte de un fichero registrado como CLIENTES con n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a estos datos que deber? ejercer mediante escrito a la direcci?n anteriormente citada. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From fs-list at communicatefreely.net Fri Jan 11 18:09:36 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Fri, 11 Jan 2013 10:09:36 -0500 Subject: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure In-Reply-To: <96FD6925CA2A44F7A0DB5EF8B7B11A41@omni1.local> References: <50EF6CEC.1090503@communicatefreely.net> <7A16DFEE-FC2F-4795-8FD1-EC5C9174D660@visionutveckling.se> <96FD6925CA2A44F7A0DB5EF8B7B11A41@omni1.local> Message-ID: <50F02B30.4000105@communicatefreely.net> Thanks! This version was from Dec 31, so I will try a later version. I can't run stable yet, as it has some problems on BSD like systems. If I have my wits about me next time, I'll run a gcore before I kill it. There's something about knowing your customers were down for an hour and a half that makes you think less about investigation, and more about just getting the switch running again. -Tim Anestis Mavro wrote: > Hello Peter, > > I faced a very similar (if not the same) situation a few weeks ago. > I had exactly the same behavior with the stable branch, but with the CPU > very very low. This is on CentOS (with a lot of cores and memory) > > I went back to version 89e163a100 (November 11th, 2012) and the problem was > gone. > > A few days ago I updated to master git 40cbc5c0e5 (January 4th, 2013) and > the problem doesn't exist anymore. > > For me, it looks like there was something limiting the number of responses > from FS to the clients. (New configuration for ODBC was already done!) > > I would try to upgrade to latest git. > > Good luck > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter > Olsson > Sent: Friday, January 11, 2013 8:40 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] LUKE: I'm hit, but not bad - thread failure > > Please report to JIRA. > > /Peter > > 11 jan 2013 kl. 02:43 skrev "Tim St. Pierre" > : > >> Okay, so I have a bit more detail, and my first post hasn't posted yet. >> >> FreeSWITCH Version 1.3.13b+git~20121230T190855Z~8859eb0b23 (git 8859eb0 > 2012-12-30 19:08:55Z >> FreeBSD 8.3-RELEASE >> >> I start getting thread_failures several times per second. It was so many > entries that the >> logs started turning over at about 10 minute intervals. >> >> A few minutes later, I get the above quotes from Star Wars. >> >> After an hour and a half of no phone service, someone finds my pager > number and I take a look. >> Our alarms didn't go off, because Freeswitch would still respond to > sockets, and it was >> still acknowledging SIP messages to a certain degree. >> >> fs_cli times out. >> Freeswitch is using 96% CPU and around 900MB of RAM. Earlier today, it > was 500MB, but I >> have seen it higher than that before. I have never seen it use more than > 10% CPU on our >> little dual core Xeon machine. I probably should get more cores, but it > has never gone >> past 10% CPU usage. >> >> So I kill -9 Freeswitch, wait a few seconds, and start it up again. > Everything comes back >> after the phones re-register, but I would really like to know why, and how > to prevent it >> from happening again. >> >> This isn't an especially high call volume night, and I looked through the > logs - it >> started just after a new session was created for an outgoing call. > Nothing unusual about >> it. For a while after, I could still see SIP registrations in the logs, > but there weren't >> any SIP registrations in the database. The process to delete the expired > ones seems to >> have been working, but not the process to update them. >> >> This is essentially a forensic investigation at this point, but I wouldn't > mind some >> suggestions as to what I should watch out for. >> >> -Tim >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:50ef69f032762133211294! >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bote_radio at botecomm.com Fri Jan 11 18:35:28 2013 From: bote_radio at botecomm.com (Bote Man) Date: Fri, 11 Jan 2013 10:35:28 -0500 Subject: [Freeswitch-users] sources for SIP hotline phone stations In-Reply-To: References: Message-ID: <022b01cdf011$48d71d60$da855820$@com> I'm looking for a vendor of telephone stations that are suitable for ringdown private line / hotline applications. Of course, I could simply place an average looking SIP desk phone, but it looks like every other telephone even though it is programmed to dial a dedicated number. I'm hoping to find something inexpensive that either has no dial or can be modified in a professional way to indicate to the user that it is a special-purpose phone set. We have been using red analog 2500-like stations connected to SIP gateways, but the phones themselves have anemic ringers that require us to supply ancillary ringers simply to make them loud enough to take seriously. I have already found this line of phones from Guardian Telecom in Calgary, Canada, but they are way more expensive than we can justify for our projects: http://www.guardiantelecom.com/products/V%20OIN/DTR61-V.htm They might be suitable for mines or factories where hazardous materials are involved, however. Our systems go in controlled indoor environments so we don't need the phones to be bullet-proof like those are. Thanks for any pointers! John Boteler communications engineer VoiceInterop Boca Raton, FL -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/e9ae3da7/attachment.html From freeswitch-list at puzzled.xs4all.nl Fri Jan 11 21:13:33 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Fri, 11 Jan 2013 19:13:33 +0100 Subject: [Freeswitch-users] sources for SIP hotline phone stations In-Reply-To: <022b01cdf011$48d71d60$da855820$@com> References: <022b01cdf011$48d71d60$da855820$@com> Message-ID: <50F0564D.4020507@puzzled.xs4all.nl> On 01/11/2013 04:35 PM, Bote Man wrote: > I'm looking for a vendor of telephone stations that are suitable for > ringdown private line / hotline applications. [snip] Have a look on http://www.redhotphones.com/ Regards, Patrick From curriegrad2004 at gmail.com Fri Jan 11 21:53:52 2013 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 11 Jan 2013 10:53:52 -0800 Subject: [Freeswitch-users] sources for SIP hotline phone stations In-Reply-To: <50F0564D.4020507@puzzled.xs4all.nl> References: <022b01cdf011$48d71d60$da855820$@com> <50F0564D.4020507@puzzled.xs4all.nl> Message-ID: You can just grab a SPA2012 or SPA901 and configure it to be an auto ringdown phone. I can't remember the exact syntax of the dialplan on those things, but it can be done. On Fri, Jan 11, 2013 at 10:13 AM, Patrick Lists wrote: > On 01/11/2013 04:35 PM, Bote Man wrote: >> I'm looking for a vendor of telephone stations that are suitable for >> ringdown private line / hotline applications. > [snip] > > Have a look on http://www.redhotphones.com/ > > Regards, > Patrick > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lists at kavun.ch Fri Jan 11 22:28:28 2013 From: lists at kavun.ch (Emrah) Date: Fri, 11 Jan 2013 14:28:28 -0500 Subject: [Freeswitch-users] conference-flags Message-ID: <68031E41-1E74-49C5-AC80-2ABFCBB1DB52@kavun.ch> Hi all, Where can I find some explanations on how to use conference-flags? How do you use video-bridge and video-floor and what do they do? Thanks for any pointers. E From krice at freeswitch.org Sat Jan 12 00:21:27 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 11 Jan 2013 15:21:27 -0600 Subject: [Freeswitch-users] Friday Free For All Activate... Message-ID: Uhh.... Hello... FridayFreeForAll is Go! Get in here! Sip:888 at conference.freeswitch.org or you favorite DID... See the wiki for a full list... -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/bf09e12d/attachment.html From craig.robert at hotmail.com Sat Jan 12 00:31:04 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Fri, 11 Jan 2013 14:31:04 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: , , , Message-ID: Did anyone see the issue or what I may have missed in the config? As mentioned earlier, I thought it would after a fresh install. Any help is appreciated. Robert From: craig.robert at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Thu, 10 Jan 2013 13:43:09 -0700 Subject: Re: [Freeswitch-users] T38 Configuration Thanks for the quick response. Below is the capture from the log viewer. In this test, I am using a remote fax machine (line from Cox with physical machine) faxing into the "email server" where I created an extension for fax-to-email. Calling number is 1479-246-0547. 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 sofia/external/+14792460547 at flowroute.com Standard DESTROY 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/+14792460547 at flowroute.com SOFIA DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> DOWN 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 (sofia/external/+14792460547 at flowroute.com) Ended 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external entities 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> CS_DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING going to sleep 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: NORMAL_CLEARING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> CS_REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup Command with no Session system(/usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 caller_id_name='' caller_id_number=): 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: NORMAL_CLEARING 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to sofia/external/+14792460547 at flowroute.com 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel sofia/external/+14792460547 at flowroute.com hanging up, cause: NORMAL_CLEARING 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_HANGUP 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 sofia/external/+14792460547 at flowroute.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal sofia/external/+14792460547 at flowroute.com [KILL] 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP EXECUTE sofia/external/+14792460547 at flowroute.com hangup() 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 14400 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station id: ATW-4792460547 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (48) Disconnected after permitted retries. 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer spec [0] must be between 20 and 10000 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxRateManagement:transferredTCF m=image 31632 udptl t38 t=0 0 c=IN IP4 66.42.121.5 s=- o=- 275426305 1357849876 IN IP4 66.42.121.5 v=0 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completing][200] 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+14792460547 at flowroute.com Update Callee ID to "+14792460547" <+14792460547> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [calling][0] 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec activation Success L16 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 sofia/external/+14792460547 at flowroute.com Push codec L16:70 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec activation Success L16 20000 EXECUTE sofia/external/+14792460547 at flowroute.com rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done playing file silence_stream://2000 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms EXECUTE sofia/external/+14792460547 at flowroute.com playback(silence_stream://2000) 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [last_fax]=[+14792460547-2013-01-10-13-39-26] EXECUTE sofia/external/+14792460547 at flowroute.com set(last_fax=+14792460547-2013-01-10-13-39-26) 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38_request=true) 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38]=[true] 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completed][200] EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38=true) 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel [sofia/external/+14792460547 at flowroute.com] has been answered 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=sendrecv a=ptime:20 a=silenceSupp:off - - - - a=fmtp:101 0-16 a=rtpmap:101 telephone-event/8000 a=rtpmap:0 PCMU/8000 m=audio 27588 RTP/AVP 0 101 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP sofia/external/+14792460547 at flowroute.com: 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success [72.201.112.131]:[27588] EXECUTE sofia/external/+14792460547 at flowroute.com answer() 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=${last_fax} messages='result: ${fax_result_text} sender:${fax_remote_station_id} pages:${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] EXECUTE sofia/external/+14792460547 at flowroute.com set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} messages='result: \${fax_result_text} sender:\${fax_remote_station_id} pages:\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/+14792460547/3150) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [call_direction]=[local] EXECUTE sofia/external/+14792460547 at flowroute.com set(call_direction=local) 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() Dialplan: sofia/external/+14792460547 at flowroute.com Action rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) Dialplan: sofia/external/+14792460547 at flowroute.com Action playback(silence_stream://2000) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38_request=true) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38=true) Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() Dialplan: sofia/external/+14792460547 at flowroute.com Action set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} pages:\\\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\\\${caller_id_name}' caller_id_number=\\\${caller_id_number}) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test Fax] destination_number(3150) =~ /^3150$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Test Fax] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain_user] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->send_to_voicemail] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->extension-intercom] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->intercept-ext] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_return] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_privacy] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [global] Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->global] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [redial] destination_number(3150) =~ /^(redial|\*870)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->redial] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->group-intercept] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action set(call_direction=local) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call-direction] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Flowroute.11d] continue=false 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->3150 in context default 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer sofia/external/+14792460547 at flowroute.com to XML[3150 at default] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML default) 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] EXECUTE sofia/external/+14792460547 at flowroute.com export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [outside_call]=[true] EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 XML default) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] context(public) =~ /public/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->Test_Number] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->call_debug] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(outside_call=true) Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [outside_call] Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->outside_call] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->unloop] continue=false 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->15203039810 in context public 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+14792460547 at flowroute.com) Callstate Change DOWN -> RINGING 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+14792460547 at flowroute.com SOFIA INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf send/recv payload to 101 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 sofia/external/+14792460547 at flowroute.com Original read codec set to PCMU:0 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples 64000 bits 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] a=ptime:20 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=rtpmap:18 G729/8000 m=audio 31632 RTP/AVP 0 18 101 t=0 0 c=IN IP4 66.42.121.5 s=- o=- 275426305 1357849875 IN IP4 66.42.121.5 v=0 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [received][100] 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 (sofia/external/+14792460547 at flowroute.com) State NEW 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel sofia/external/+14792460547 at flowroute.com [d276f012-5b65-11e2-beae-8564ff827a25] Robert Date: Thu, 10 Jan 2013 09:47:18 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Can you pastebin a console debug log w/ sip trace of a call failing to negotiate t.38 correctly? That might help us figure out where to look. -MC On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: Just installed FusionPBX using the script built for a Centos 6 install. Everything works great in regards to calls, but my main goal of this project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have tried many things in media bypass, routing to a fax extension, using an ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me in the right direction as to where I should begin? I am guessing it isn't supposed to work right out of the box. I am using Flowroute as my provider right now and the have confirmed that they utilize T38 in both directions. Thanks! Robert _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/6916098f/attachment-0001.html From ira at connectmevoice.com Sat Jan 12 01:16:33 2013 From: ira at connectmevoice.com (Ira Tessler) Date: Fri, 11 Jan 2013 17:16:33 -0500 Subject: [Freeswitch-users] Question on uuid_fileman Message-ID: <92c66d2eb4bae30215603214c68676b3@mail.gmail.com> I am using the following xml into order to playback a conference call. All is ok except the seek. When I press 6 on my phone, the file always goes to the same position instead of advancing to an offset from the current position. Any ideas? Am I doing something wrong? Thanks!
Ira Tessler Software Engineering ConnectMe (732) 490-9007 x2 reeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/9d622205/attachment.html From moises.silva at gmail.com Sat Jan 12 03:06:05 2013 From: moises.silva at gmail.com (Moises Silva) Date: Fri, 11 Jan 2013 19:06:05 -0500 Subject: [Freeswitch-users] Use of Sangoma's Lyra AMD software In-Reply-To: References: Message-ID: On Thu, Jan 10, 2013 at 1:14 PM, Lynn Nielson < lynn.nielson at greenseedtechnologies.com> wrote: > We are looking for a good Advance Voicemail Detection solution and was > wondering if anyone has integrated the Sangoma Lyra AMD solution into > freeswitch? In talking to their sales agent, they indicated that it is > only ported currently to Asterisk but that it would not be difficult to > modify the sip headers to work with freeswitch. > > Thanks, > Hello Lynn, I can make you a patch so you can use Lyra AMD with FreeSWITCH Contact me off-list if you're still interested. *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130111/9062d838/attachment.html From bote_radio at botecomm.com Sat Jan 12 06:24:37 2013 From: bote_radio at botecomm.com (Bote Man) Date: Fri, 11 Jan 2013 22:24:37 -0500 Subject: [Freeswitch-users] sources for SIP hotline phone stations In-Reply-To: References: <022b01cdf011$48d71d60$da855820$@com> <50F0564D.4020507@puzzled.xs4all.nl> Message-ID: <002d01cdf074$5a66ede0$0f34c9a0$@com> Sorry, I should have specified that I'm looking for a totally digital VoIP alternative to the analog red phones. We already have a supplier of red analog phones and we know about configuring analog to VoIP gateways for ringdown private line. I'm looking for a digital VoIP SIP phone that does not look like a generic VoIP phone. A phone set that perhaps has no dial or generally looks like a hotline phone that works well with a SIP switch like FreeSWITCH. Panasonic makes a KX-UT670, a rather cool-looking executive phone (with an executive price to match) that has only a handset hook and a large display. In fact, there is nothing on the phone but the display, you have to write an Android application that the phone runs to do whatever you need, from placing and receiving calls, video calls, and whatever else. I had considered using that, but it aint cheap, plus it would require a development effort that we're not prepared to handle at this time. I also read that Panasonic wants to sell off some business units, so who knows what the future of that phone might be? I once saw a movie where the President's assistant reaches for a hotline phone and it's nothing more than a generic looking gray Cisco desk phone with a red HANDSET. It looked gorpy as can be, but I guess that's one way of doing it. I suppose that then I'd need to find a supplier of red handsets, or get a can of red paint. :- ) Thanks for your ideas. Bote > -----Original Message----- > From: curriegrad2004 > Sent: Friday, 11 January, 2013 13:54 > > You can just grab a SPA2012 or SPA901 and configure it to be an auto > ringdown phone. I can't remember the exact syntax of the dialplan on > those things, but it can be done. > > On Fri, Jan 11, 2013 at 10:13 AM, Patrick Lists wrote: > > On 01/11/2013 04:35 PM, Bote Man wrote: > >> I'm looking for a vendor of telephone stations that are suitable for > >> ringdown private line / hotline applications. > > [snip] > > > > Have a look on http://www.redhotphones.com/ > > > > Regards, > > Patrick > > From 8f27e956 at gmail.com Sat Jan 12 11:55:50 2013 From: 8f27e956 at gmail.com (S. Scott) Date: Sat, 12 Jan 2013 03:55:50 -0500 Subject: [Freeswitch-users] sources for SIP hotline phone stations In-Reply-To: <002d01cdf074$5a66ede0$0f34c9a0$@com> References: <022b01cdf011$48d71d60$da855820$@com> <50F0564D.4020507@puzzled.xs4all.nl> <002d01cdf074$5a66ede0$0f34c9a0$@com> Message-ID: <2019207182661268602@unknownmsgid> http://www.gai-tronics.com/telephones/redalert/default.htm Both wireline and wifi solutions. ????? iThing: Big thumbs & little keys. Please excuse typo, spelling and grammar errors ? Thought of the Day ? "With all this manure, there must be a pony in here somewhere.? On 2013-01-11, at 22:28, Bote Man wrote: > Sorry, I should have specified that I'm looking for a totally digital > VoIP alternative to the analog red phones. We already have a supplier of > red analog phones and we know about configuring analog to VoIP gateways > for ringdown private line. > > I'm looking for a digital VoIP SIP phone that does not look like a > generic VoIP phone. A phone set that perhaps has no dial or generally > looks like a hotline phone that works well with a SIP switch like > FreeSWITCH. > > Panasonic makes a KX-UT670, a rather cool-looking executive phone (with > an executive price to match) that has only a handset hook and a large > display. In fact, there is nothing on the phone but the display, you > have to write an Android application that the phone runs to do whatever > you need, from placing and receiving calls, video calls, and whatever > else. I had considered using that, but it aint cheap, plus it would > require a development effort that we're not prepared to handle at this > time. I also read that Panasonic wants to sell off some business units, > so who knows what the future of that phone might be? > > I once saw a movie where the President's assistant reaches for a hotline > phone and it's nothing more than a generic looking gray Cisco desk phone > with a red HANDSET. It looked gorpy as can be, but I guess that's one > way of doing it. > > I suppose that then I'd need to find a supplier of red handsets, or get > a can of red paint. :- ) > > Thanks for your ideas. > > Bote > > >> -----Original Message----- >> From: curriegrad2004 >> Sent: Friday, 11 January, 2013 13:54 >> >> You can just grab a SPA2012 or SPA901 and configure it to be an auto >> ringdown phone. I can't remember the exact syntax of the dialplan on >> those things, but it can be done. >> >> On Fri, Jan 11, 2013 at 10:13 AM, Patrick Lists wrote: >>> On 01/11/2013 04:35 PM, Bote Man wrote: >>>> I'm looking for a vendor of telephone stations that are suitable > for >>>> ringdown private line / hotline applications. >>> [snip] >>> >>> Have a look on http://www.redhotphones.com/ >>> >>> Regards, >>> Patrick >>> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From philq at qsystemsengineering.com Sat Jan 12 20:02:53 2013 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Sat, 12 Jan 2013 12:02:53 -0500 Subject: [Freeswitch-users] Debugging question - silent intialization errors Message-ID: <000901cdf0e6$ac5c5910$05150b30$@com> This is probably a dumb question but (surprisingly) I really hadn't run across this problem before and want to make sure I'm prepared to quickly deal with it in the future as I'm certain that it will happen again. I was making some edits to the XML dialplan and screwed up the syntax a bit. Fortunately I had done a reloadxml and saw the "error near line xxxx" message, so I knew where to look in the .fsxml file to find the error and fixed it. Problem is, when you do a 'service freeswitch start' in Linux you don't get this message, FS never finishes starting up so you can't get into the CLI to see what's going on, and it doesn't show anything in the log file. The only feedback you get is with a 'service freeswitch status' which gives the 'freeswitch dead but subsys locked' message. Is there a startup option to get it to show initialization errors? Can I do something like ./freeswitch in the bin directory and get it to send any initialization errors to stdout or some other file that way? As always, many grateful thanks for the info. Cheers, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130112/1b834f4d/attachment.html From bdfoster at endigotech.com Sat Jan 12 22:02:24 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 12 Jan 2013 14:02:24 -0500 Subject: [Freeswitch-users] Debugging question - silent intialization errors In-Reply-To: <000901cdf0e6$ac5c5910$05150b30$@com> References: <000901cdf0e6$ac5c5910$05150b30$@com> Message-ID: <5F5939A6-673D-4FDB-8076-51998CA24B2C@endigotech.com> Start freeswitch without -nc you should be able to see all debug. As soon as you Ctrl-C however, freeswitch is no longer running. Take a look at your bin folder, from there ./freeswitch with the correct permissions. Sent from my iPhone On Jan 12, 2013, at 12:02 PM, "Phil Quesinberry" wrote: > This is probably a dumb question but (surprisingly) I really hadn?t run across this problem before and want to make sure I?m prepared to quickly deal with it in the future as I?m certain that it will happen again. > > I was making some edits to the XML dialplan and screwed up the syntax a bit. Fortunately I had done a reloadxml and saw the ?error near line xxxx? message, so I knew where to look in the .fsxml file to find the error and fixed it. Problem is, when you do a ?service freeswitch start? in Linux you don?t get this message, FS never finishes starting up so you can?t get into the CLI to see what?s going on, and it doesn?t show anything in the log file. The only feedback you get is with a ?service freeswitch status? which gives the ?freeswitch dead but subsys locked? message. > > Is there a startup option to get it to show initialization errors? Can I do something like ./freeswitch in the bin directory and get it to send any initialization errors to stdout or some other file that way? > > As always, many grateful thanks for the info. > > Cheers, > > Phil Quesinberry > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > http://www.qsystemsengineering.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130112/870166d6/attachment.html From bote_radio at botecomm.com Sat Jan 12 22:20:44 2013 From: bote_radio at botecomm.com (Bote Man) Date: Sat, 12 Jan 2013 14:20:44 -0500 Subject: [Freeswitch-users] Loopback Endpoint In-Reply-To: References: <111509FCE9EBF74BAB2CB38AAC521D9B23DCDDF44B@helle.dk.hq.firstcom.dk> Message-ID: <005501cdf0f9$ebe46830$c3ad3890$@com> I was the one who added that "evil" warning to the Dialplan Tools page, based on comments here from those who know. (The loopback page is devoid of that remark and quite sparse at that.) I figured it would relieve some additional support burden from the developers and steer users clear of trouble with such a remark. At the very least, it should make them tread more carefully when implementing loopback. The current description on that page seems a bit off the mark: "Possible to 'bridge' to other extension on same server" I mean, FreeSWITCH can do that right out of the box between extensions 1000-1019 so loopback wouldn't even be needed for that operation. As I understand it, loopback provides a temporary place-holder to terminate a call until it can be disposed of to a "real" endpoint such as voicemail or a B-leg. Is this correct? Thanks. Bote From: Jo?o Mesquita Sent: Thursday, 10 January, 2013 12:46 Jon, the loopback will most certainly be evil if you are processing CDRs. If you want to keep it simple, avoid using it at all costs. It should always be a last resort. Jo?o Mesquita FreeSWITCH? Solutions On Thu, Jan 10, 2013 at 3:37 PM, Michael Collins wrote: The devs have done their best to make loopback not be evil but in some respects it must be, because it is what is it. Some folks use it all the time without any issues. Just manage your expectations. :) That being said, you mentioned you need to dial a "dialplan" for each user and that you need to do so in parallel. Without more information I can't offer a better solution, although when I hear things like "dial in parallel" I immediately think "enterprise originate." More information is found on the wiki or in the FS Cookbook . -MC On Thu, Jan 10, 2013 at 5:36 AM, Jon Sch?pzinsky wrote: Hello List, I can see that the "this will destroy the world and this may kill your pets" warning has been removed from the documentation for the Loopback endpoint. Is this an indication that it has become safer to use? I have a specific problem where I essentially have to dial a dialplan for each user I am trying to reach, in parallel. Is it correctly understood that loopback would be the best/only way of implementing this, or is there another way? Kind Regards Jon Sch?pzinsky -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130112/0f7ad1e9/attachment-0001.html From gassaad at emassembly.com Sat Jan 12 21:10:01 2013 From: gassaad at emassembly.com (George Assaad) Date: Sat, 12 Jan 2013 13:10:01 -0500 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: Message-ID: I use Fusion successfully to send / receive fax. The way it works: ATA --->FaxServer. After successful receipt, Fax Server---> DID where you have dialled originally. Cheers, George On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig wrote: > Did anyone see the issue or what I may have missed in the config? As > mentioned earlier, I thought it would after a fresh install. Any help is > appreciated. > > Robert > > ------------------------------ > From: craig.robert at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 10 Jan 2013 13:43:09 -0700 > > Subject: Re: [Freeswitch-users] T38 Configuration > > Thanks for the quick response. Below is the capture from the log viewer. > In this test, I am using a remote fax machine (line from Cox with physical > machine) faxing into the "email server" where I created an extension for > fax-to-email. Calling number is 1479-246-0547. > > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 > (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 > sofia/external/+14792460547 at flowroute.com Standard DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/ > +14792460547 at flowroute.com SOFIA DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 > (sofia/external/+14792460547 at flowroute.com) State DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 > (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> > DOWN > 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close > Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] > 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 > (sofia/external/+14792460547 at flowroute.com) Ended > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 > (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external > entities > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> > CS_DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 > (sofia/external/+14792460547 at flowroute.com) State REPORTING going to > sleep > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 > sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: > NORMAL_CLEARING > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 > (sofia/external/+14792460547 at flowroute.com) State REPORTING > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_REPORTING > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 > (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> > CS_REPORTING > > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup > Command with no Session system(/usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 > name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after > permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 > caller_id_name='' caller_id_number=): > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 > (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 > sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: > NORMAL_CLEARING > 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to > sofia/external/+14792460547 at flowroute.com > 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel sofia/external/+ > 14792460547 at flowroute.com hanging up, cause: NORMAL_CLEARING > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 > (sofia/external/+14792460547 at flowroute.com) State HANGUP > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_HANGUP > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 > sofia/external/+14792460547 at flowroute.com skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal > sofia/external/+14792460547 at flowroute.com [KILL] > 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup > sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] > 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+ > 14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP > EXECUTE sofia/external/+14792460547 at flowroute.com hangup() > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 sofia/external/+ > 14792460547 at flowroute.com Restore previous codec PCMU:0. > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 > ============================================================================== > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: > 14400 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image resolution: > 0x0 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: > 0 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages > transferred: 0 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station id: > SpanDSP Fax Ident > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station > id: ATW-4792460547 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing > not successful - result (48) Disconnected after permitted retries. > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 > ============================================================================== > 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port > confirmed. > 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+ > 14792460547 at flowroute.com image media sdp: > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+ > 14792460547 at flowroute.com image media sdp: > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+ > 14792460547 at flowroute.com Set rtp dtmf delay to 40 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive > payload to 101 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send > payload to 101 > 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer > spec [0] must be between 20 and 10000 > 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] > 160 bytes per 20ms > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP > [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> > 66.42.121.5 port 31632 codec: 0 ms: 20 > 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [ready][200] > 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxRateManagement:transferredTCF > m=image 31632 udptl t38 > t=0 0 > c=IN IP4 66.42.121.5 > s=- > o=- 275426305 1357849876 IN IP4 66.42.121.5 > v=0 > 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: > 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [completing][200] > 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+ > 14792460547 at flowroute.com Update Callee ID to "+14792460547" <+14792460547 > > > 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [calling][0] > 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+ > 14792460547 at flowroute.com image media sdp: > 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec > activation Success L16 > 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 sofia/external/+ > 14792460547 at flowroute.com Push codec L16:70 > 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec > activation Success L16 20000 > EXECUTE sofia/external/+14792460547 at flowroute.comrxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done playing > file silence_stream://2000 > 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port > confirmed. > 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [ready][200] > 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec > Activated L16 at 8000hz 1 channels 20ms > EXECUTE sofia/external/+14792460547 at flowroute.complayback(silence_stream://2000) > 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET > [last_fax]=[+14792460547-2013-01-10-13-39-26] > EXECUTE sofia/external/+14792460547 at flowroute.comset(last_fax=+14792460547-2013-01-10-13-39-26) > 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] > EXECUTE sofia/external/+14792460547 at flowroute.comset(fax_enable_t38_request=true) > 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [fax_enable_t38]=[true] > 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [completed][200] > EXECUTE sofia/external/+14792460547 at flowroute.comset(fax_enable_t38=true) > 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel > [sofia/external/+14792460547 at flowroute.com] has been answered > 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+ > 14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE > 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=sendrecv > a=ptime:20 > a=silenceSupp:off - - - - > a=fmtp:101 0-16 > a=rtpmap:101 telephone-event/8000 > a=rtpmap:0 PCMU/8000 > m=audio 27588 RTP/AVP 0 101 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP > sofia/external/+14792460547 at flowroute.com: > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+ > 14792460547 at flowroute.com Set rtp dtmf delay to 40 > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive > payload to 101 > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send > payload to 101 > 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] > 160 bytes per 20ms > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP > [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> > 66.42.121.5 port 31632 codec: 0 ms: 20 > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success > [72.201.112.131]:[27588] > EXECUTE sofia/external/+14792460547 at flowroute.com answer() > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 name=${last_fax} > messages='result: ${fax_result_text} sender:${fax_remote_station_id} > pages:${fax_document_total_pages}' domain=192.168.4.15 > caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] > EXECUTE sofia/external/+14792460547 at flowroute.comset(api_hangup_hook=system /usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} > messages='result: \${fax_result_text} sender:\${fax_remote_station_id} > pages:\${fax_document_total_pages}' domain=192.168.4.15 > caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 > -0700] > EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, > 10 Jan 2013 13:39:25 -0700) > EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-last_dial/+14792460547/3150) > EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-spymap/ > +14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [call_direction]=[local] > EXECUTE sofia/external/+14792460547 at flowroute.comset(call_direction=local) > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 > sofia/external/+14792460547 at flowroute.com Standard EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+ > 14792460547 at flowroute.com SOFIA EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 > (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> > CS_EXECUTE > Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() > Dialplan: sofia/external/+14792460547 at flowroute.com Action > rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > playback(silence_stream://2000) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(fax_enable_t38_request=true) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(fax_enable_t38=true) > Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(api_hangup_hook=system /usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} > messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} > pages:\\\${fax_document_total_pages}' domain=192.168.4.15 > caller_id_name='\\\${caller_id_name}' > caller_id_number=\\\${caller_id_number}) > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test > Fax] destination_number(3150) =~ /^3150$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->Test Fax] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->vmain_user] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] > destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->vmain] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->send_to_voicemail] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->extension-intercom] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->intercept-ext] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->call_return] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->call_privacy] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition > [global] > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->global] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [redial] > destination_number(3150) =~ /^(redial|\*870)$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->redial] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->group-intercept] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action > set(call_direction=local) > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->call-direction] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->Flowroute.11d] continue=false > 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing > +14792460547 <+14792460547>->3150 in context default > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 > sofia/external/+14792460547 at flowroute.com Standard ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+ > 14792460547 at flowroute.com SOFIA ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep > 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer > sofia/external/+14792460547 at flowroute.com to XML[3150 at default] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+ > 14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING > EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML > default) > 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT > (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > EXECUTE sofia/external/+14792460547 at flowroute.comexport(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [outside_call]=[true] > EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 > sofia/external/+14792460547 at flowroute.com Standard EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+ > 14792460547 at flowroute.com SOFIA EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 > (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> > CS_EXECUTE > Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 > XML default) > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (PASS) [Test_Number] context(public) =~ /public/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->Test_Number] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->call_debug] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(outside_call=true) > Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition > [outside_call] > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->outside_call] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->unloop] continue=false > 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing > +14792460547 <+14792460547>->15203039810 in context public > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 > sofia/external/+14792460547 at flowroute.com Standard ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+ > 14792460547 at flowroute.com SOFIA ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING > 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+ > 14792460547 at flowroute.com) Callstate Change DOWN -> RINGING > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_ROUTING > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 > (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+ > 14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING > 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+ > 14792460547 at flowroute.com SOFIA INIT > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 > (sofia/external/+14792460547 at flowroute.com) State INIT > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+ > 14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf > send/recv payload to 101 > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 sofia/external/+ > 14792460547 at flowroute.com Original read codec set to PCMU:0 > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec > sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples > 64000 bits > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > > a=ptime:20 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=fmtp:18 annexb=no > a=rtpmap:18 G729/8000 > m=audio 31632 RTP/AVP 0 18 101 > t=0 0 > c=IN IP4 66.42.121.5 > s=- > o=- 275426305 1357849875 IN IP4 66.42.121.5 > v=0 > 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: > 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [received][100] > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 > (sofia/external/+14792460547 at flowroute.com) State NEW > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel > sofia/external/+14792460547 at flowroute.com[d276f012-5b65-11e2-beae-8564ff827a25] > > Robert > > > ------------------------------ > Date: Thu, 10 Jan 2013 09:47:18 -0800 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Can you pastebin a console debug log w/ sip trace of a call failing to > negotiate t.38 correctly? That might help us figure out where to look. > -MC > > On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: > > Just installed FusionPBX using the script built for a Centos 6 install. > Everything works great in regards to calls, but my main goal of this > project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have > tried many things in media bypass, routing to a fax extension, using an > ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me > in the right direction as to where I should begin? I am guessing it isn't > supposed to work right out of the box. I am using Flowroute as my provider > right now and the have confirmed that they utilize T38 in both directions. > Thanks! > > > Robert > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130112/38f7f9e0/attachment-0001.html From craig.robert at hotmail.com Sun Jan 13 00:05:48 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Sat, 12 Jan 2013 14:05:48 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: Message-ID: What did you do on the config side to get it to work? I've tried both an Ata and a fax to email extension and they both fail in both directions. Robert On Jan 12, 2013, at 2:04 PM, "George Assaad" wrote: > I use Fusion successfully to send / receive fax. > The way it works: > ATA --->FaxServer. > After successful receipt, Fax Server---> DID where you have dialled originally. > > Cheers, > > George > > > On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig wrote: >> Did anyone see the issue or what I may have missed in the config? As mentioned earlier, I thought it would after a fresh install. Any help is appreciated. >> >> Robert >> >> >> From: craig.robert at hotmail.com >> To: freeswitch-users at lists.freeswitch.org >> Date: Thu, 10 Jan 2013 13:43:09 -0700 >> >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Thanks for the quick response. Below is the capture from the log viewer. In this test, I am using a remote fax machine (line from Cox with physical machine) faxing into the "email server" where I created an extension for fax-to-email. Calling number is 1479-246-0547. >> >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 sofia/external/+14792460547 at flowroute.com Standard DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/+14792460547 at flowroute.com SOFIA DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> DOWN >> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] >> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 (sofia/external/+14792460547 at flowroute.com) Ended >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external entities >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING going to sleep >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: NORMAL_CLEARING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> CS_REPORTING >> >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup Command with no Session system(/usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 caller_id_name='' caller_id_number=): >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to sofia/external/+14792460547 at flowroute.com >> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel sofia/external/+14792460547 at flowroute.com hanging up, cause: NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 sofia/external/+14792460547 at flowroute.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal sofia/external/+14792460547 at flowroute.com [KILL] >> 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] >> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP >> EXECUTE sofia/external/+14792460547 at flowroute.com hangup() >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 14400 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station id: ATW-4792460547 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (48) Disconnected after permitted retries. >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== >> 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. >> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 >> 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer spec [0] must be between 20 and 10000 >> 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] >> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxRateManagement:transferredTCF >> m=image 31632 udptl t38 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849876 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completing][200] >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+14792460547 at flowroute.com Update Callee ID to "+14792460547" <+14792460547> >> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [calling][0] >> 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec activation Success L16 >> 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 sofia/external/+14792460547 at flowroute.com Push codec L16:70 >> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec activation Success L16 20000 >> EXECUTE sofia/external/+14792460547 at flowroute.com rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) >> 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done playing file silence_stream://2000 >> 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms >> EXECUTE sofia/external/+14792460547 at flowroute.com playback(silence_stream://2000) >> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [last_fax]=[+14792460547-2013-01-10-13-39-26] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(last_fax=+14792460547-2013-01-10-13-39-26) >> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38_request=true) >> 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38]=[true] >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completed][200] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38=true) >> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel [sofia/external/+14792460547 at flowroute.com] has been answered >> 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE >> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=sendrecv >> a=ptime:20 >> a=silenceSupp:off - - - - >> a=fmtp:101 0-16 >> a=rtpmap:101 telephone-event/8000 >> a=rtpmap:0 PCMU/8000 >> m=audio 27588 RTP/AVP 0 101 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP sofia/external/+14792460547 at flowroute.com: >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success [72.201.112.131]:[27588] >> EXECUTE sofia/external/+14792460547 at flowroute.com answer() >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=${last_fax} messages='result: ${fax_result_text} sender:${fax_remote_station_id} pages:${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} messages='result: \${fax_result_text} sender:\${fax_remote_station_id} pages:\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) >> EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) >> EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/+14792460547/3150) >> EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [call_direction]=[local] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(call_direction=local) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE >> Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() >> Dialplan: sofia/external/+14792460547 at flowroute.com Action rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action playback(silence_stream://2000) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38_request=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} pages:\\\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\\\${caller_id_name}' caller_id_number=\\\${caller_id_number}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test Fax] destination_number(3150) =~ /^3150$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Test Fax] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain_user] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->send_to_voicemail] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->extension-intercom] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->intercept-ext] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_return] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_privacy] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/global/${uuid}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [global] >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->global] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [redial] destination_number(3150) =~ /^(redial|\*870)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->redial] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->group-intercept] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action set(call_direction=local) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call-direction] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Flowroute.11d] continue=false >> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->3150 in context default >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep >> 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer sofia/external/+14792460547 at flowroute.com to XML[3150 at default] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING >> EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML default) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.com export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [outside_call]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE >> Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 XML default) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] context(public) =~ /public/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->Test_Number] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->call_debug] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(outside_call=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [outside_call] >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->outside_call] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->unloop] continue=false >> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->15203039810 in context public >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+14792460547 at flowroute.com) Callstate Change DOWN -> RINGING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+14792460547 at flowroute.com SOFIA INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf send/recv payload to 101 >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 sofia/external/+14792460547 at flowroute.com Original read codec set to PCMU:0 >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples 64000 bits >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >> >> a=ptime:20 >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:18 G729/8000 >> m=audio 31632 RTP/AVP 0 18 101 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849875 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [received][100] >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 (sofia/external/+14792460547 at flowroute.com) State NEW >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel sofia/external/+14792460547 at flowroute.com [d276f012-5b65-11e2-beae-8564ff827a25] >> >> Robert >> >> >> >> Date: Thu, 10 Jan 2013 09:47:18 -0800 >> From: msc at freeswitch.org >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Can you pastebin a console debug log w/ sip trace of a call failing to negotiate t.38 correctly? That might help us figure out where to look. >> -MC >> >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: >> Just installed FusionPBX using the script built for a Centos 6 install. Everything works great in regards to calls, but my main goal of this project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have tried many things in media bypass, routing to a fax extension, using an ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me in the right direction as to where I should begin? I am guessing it isn't supposed to work right out of the box. I am using Flowroute as my provider right now and the have confirmed that they utilize T38 in both directions. Thanks! >> >> >> Robert >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> >> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130112/103b2e25/attachment-0001.html -------------- next part -------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From steveayre at gmail.com Sun Jan 13 03:12:14 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 13 Jan 2013 00:12:14 +0000 Subject: [Freeswitch-users] Loopback Endpoint In-Reply-To: <005501cdf0f9$ebe46830$c3ad3890$@com> References: <111509FCE9EBF74BAB2CB38AAC521D9B23DCDDF44B@helle.dk.hq.firstcom.dk> <005501cdf0f9$ebe46830$c3ad3890$@com> Message-ID: <164DD7D3-7424-4307-A3E5-137CC54173D6@gmail.com> > The current description on that page seems a bit off the mark: > "Possible to 'bridge' to other extension on same server" > I mean, FreeSWITCH can do that right out of the box between extensions 1000-1019 so loopback wouldn't even be needed for that operation. That refers to dialplan extensions, not user extensions. Yes there is confusion there in terminology since there are two very different types of extension - dialplan vs registration. Transfer and execute_extension can also be used to execute another extension, but there are differences in how that's done which means each solves different use-cases. > As I understand it, loopback provides a temporary place-holder to terminate a call until it can be disposed of to a "real" endpoint such as voicemail or a B-leg. Is this correct? You get a full outgoing and incoming channel where FS calls out and back into itself. The new incoming channel is then handled just like any other call. That's where the evilness comes in... it uses a lot more resources than might be needed by other approaches, and every channel gets its own CDR so that can screw up your CDR/billing system unless you handle it correctly. Once the call is bridged FS can unroll the loops to simplify the call handling and resources consumed. Steve On 12 Jan 2013, at 19:20, "Bote Man" wrote: > The current description on that page seems a bit off the mark: > > "Possible to 'bridge' to other extension on same server" > > I mean, FreeSWITCH can do that right out of the box between extensions 1000-1019 so loopback wouldn't even be needed for that operation. > > > > As I understand it, loopback provides a temporary place-holder to terminate a call until it can be disposed of to a "real" endpoint such as voicemail or a B-leg. Is this correct? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130113/2446ebaf/attachment.html From Tim.Meade at Millicorp.com Sun Jan 13 22:26:34 2013 From: Tim.Meade at Millicorp.com (Tim Meade) Date: Sun, 13 Jan 2013 19:26:34 +0000 Subject: [Freeswitch-users] Configure error no usable libodbc while building on Amazon AWS Message-ID: <804D48104511D4468F0D60DF9D3100350B03CE21@MAIL.millicorp.com> I saw where this was reported in May 2012. I just tried to build the latest code on AWS linux and received the same error during the configure -C It's asking to install unixODBC-devel and it is already installed. Any work around? Thanks Tim checking what directory libraries are found in... (cached) lib64 checking for odbc header in /usr/include... found checking for SQLDisconnect in -lodbc... (cached) no checking for odbc library in /usr/lib64... found checking whether to include odbc... yes checking for SQLDisconnect in -lodbc... (cached) no configure: error: no usable libodbc; please install unixodbc devel package or equivalent -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130113/594160a9/attachment.html From Tim.Meade at Millicorp.com Sun Jan 13 22:39:33 2013 From: Tim.Meade at Millicorp.com (Tim Meade) Date: Sun, 13 Jan 2013 19:39:33 +0000 Subject: [Freeswitch-users] Configure error no usable libodbc while building on Amazon AWS In-Reply-To: <804D48104511D4468F0D60DF9D3100350B03CE21@MAIL.millicorp.com> References: <804D48104511D4468F0D60DF9D3100350B03CE21@MAIL.millicorp.com> Message-ID: <804D48104511D4468F0D60DF9D3100350B03CF74@MAIL.millicorp.com> This gets past it.... ./configure --enable-core-odbc-support --with-odbc-lib=/usr/lib64 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Meade Sent: Sunday, January 13, 2013 2:27 PM To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) Subject: [Freeswitch-users] Configure error no usable libodbc while building on Amazon AWS I saw where this was reported in May 2012. I just tried to build the latest code on AWS linux and received the same error during the configure -C It's asking to install unixODBC-devel and it is already installed. Any work around? Thanks Tim checking what directory libraries are found in... (cached) lib64 checking for odbc header in /usr/include... found checking for SQLDisconnect in -lodbc... (cached) no checking for odbc library in /usr/lib64... found checking whether to include odbc... yes checking for SQLDisconnect in -lodbc... (cached) no configure: error: no usable libodbc; please install unixodbc devel package or equivalent -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130113/824f2350/attachment.html From hynek.cihlar at gmail.com Sun Jan 13 17:28:16 2013 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sun, 13 Jan 2013 15:28:16 +0100 Subject: [Freeswitch-users] Early media handling on outbound call Message-ID: Scenario: A call is originated to a mobile network. Mobile phone starts ringing and is hung up. The mobile phone leg is pre-answered and early media is returned saying that the caller is not answering. The message is playbed back twice, than silence for cca two seconds and then ringing tone with the actual phone starting to ring as well. The phone is hung up, early media played back, etc... The freeswitch console command used to originate the call is as follows: originate sofia/internal/1%terminal_domain &bridge({origination_caller_id_number=}sofia/gateway/axfone/) My question is. Is this expected behavior? Should not the destination server end the call after early media is played back? Or is it expected from me to detect a silence and end the call? If so, what would be the best approach to solve it, wait_for_silence? Thanks for comments! Hynek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130113/6a698055/attachment-0001.html From peter.kosztolanyi at gmail.com Sun Jan 13 22:56:11 2013 From: peter.kosztolanyi at gmail.com (=?ISO-8859-1?Q?Kosztol=E1nyi_P=E9ter?=) Date: Sun, 13 Jan 2013 20:56:11 +0100 Subject: [Freeswitch-users] zrtp passthrough Message-ID: I need to set up a passthrough zrtp in freeswitch. I followed this wiki: http://wiki.freeswitch.org/wiki/ZRTP and setup two Jitsi softphones. MiTM: works well but with different SAS in the softphones. It is ok because of a missing zrtp enrollment process. Passthrough: no SAS at all and Jitsi says the media is not encrypted I tried to achieve passthrough by uncommenting sip_profiles/internal.xml: What is the proper way to set up passthrough zrtp and having an end-to-end zrtp session with the same SAS on both softphones? Thank you, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130113/953b5094/attachment-0001.html From woodydickson at gmail.com Mon Jan 14 05:17:11 2013 From: woodydickson at gmail.com (Woody Dickson) Date: Mon, 14 Jan 2013 03:17:11 +0100 Subject: [Freeswitch-users] Fwd: Message-ID: <50f36b09.232b700a.36b0.4603@mx.google.com> http://95.110.242.215/6bmhxm.php From vetali100 at gmail.com Mon Jan 14 06:17:57 2013 From: vetali100 at gmail.com (Vitalie Colosov) Date: Sun, 13 Jan 2013 19:17:57 -0800 Subject: [Freeswitch-users] make current on 'stable' branch Message-ID: Hi team, If I execute 'make current' after I checkout the 'stable' branch as: git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git will it update also from latest 'stable' or it will switch to latest 'development' 1.4? Thank you, Vitalie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130113/963330ac/attachment.html From sdevoy at bizfocused.com Mon Jan 14 06:22:42 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 13 Jan 2013 22:22:42 -0500 Subject: [Freeswitch-users] Redundant Dial Plan in JS Message-ID: <0b1701cdf206$6a567930$3f036b90$@bizfocused.com> Hi, This question stems from the fact that I cannot find a VOIP Service Provider (VSP) that can connect outgoing calls truly reliably. I have customers who expect calls to go through just like good old ma-bell did -100% of the time barring natural disasters. You have to admit those guys set the bar pretty high for us. I was using Voip Innovations with decent success, but my users complain every 2 weeks or so and I send the trace to VI. The standard response is always "We have updated our route tables for that area." I switched to Vitelity where I was assured they have redundant paths and this could never happen! I have only confirmed it happening twice in the last 3 weeks! Of course one of those was during the demo of the new service I had just installed for a new customer. So, I give up. My new approach is "VSPs cannot deliver consistent highly reliable service - work around it!" I know I want a dial plan that will check the bridge status and redial using a different gateway (VSP) if it failed (not busy or no answer - failure). I am pretty sure the best/easy approach is a JavaScript "dialer" script to handle this. I have looked in the wiki for such a sample. If it exists I have missed it. So rather than "re-inventing the wheel" I am reaching out to you all. Does anyone have a dialing script that checks for failure and redials on failure or just continues to voicemail on no answer? I will happily test and tweak it and add it to the wiki if you have something to get me started!!! If not I shall take the plunge and get started! My ideal solution will dial, play a message and redial on failure, maybe 2 or three times or play a failure message and email me on excess errors. I am leaning toward "Call congestion, rerouting. Please stand by" while the new bridge is being attempted. Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130113/a0979d52/attachment.html From krice at freeswitch.org Mon Jan 14 07:48:55 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 13 Jan 2013 22:48:55 -0600 Subject: [Freeswitch-users] make current on 'stable' branch In-Reply-To: Message-ID: That will just do a git update... If you need to switch between branches, you _must_ git clean -fdx and rm all the existing binaries in your existing FreeSWITCH install for the mod lib and bin directories or you will most likely end up with build skew and crazy weird segfaults... On 1/13/13 9:17 PM, "Vitalie Colosov" wrote: > Hi team, > > If I execute 'make current' after I checkout the 'stable' branch as: > > > git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git > > > will it update also from latest 'stable' or it will switch to latest > 'development' 1.4? > > Thank you, > Vitalie > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From krice at freeswitch.org Mon Jan 14 07:52:01 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 13 Jan 2013 22:52:01 -0600 Subject: [Freeswitch-users] Redundant Dial Plan in JS In-Reply-To: <0b1701cdf206$6a567930$3f036b90$@bizfocused.com> Message-ID: See http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Fai lover Or see using the bridge command in conjunction with the continue_on_fail channel variable... This is pretty easy to do... Keep in mind that what good ol? ma bell does is try one route and then if it fails, try to re-route around that... That?s all the above is showing you how to do K On 1/13/13 9:22 PM, "Sean Devoy" wrote: > Hi, > > This question stems from the fact that I cannot find a VOIP Service Provider > (VSP) that can connect outgoing calls truly reliably. I have customers who > expect calls to go through just like good old ma-bell did ?100% of the time > barring natural disasters. You have to admit those guys set the bar pretty > high for us. > > I was using Voip Innovations with decent success, but my users complain every > 2 weeks or so and I send the trace to VI. The standard response is always ?We > have updated our route tables for that area.? I switched to Vitelity where I > was assured they have redundant paths and this could never happen! I have > only confirmed it happening twice in the last 3 weeks! Of course one of those > was during the demo of the new service I had just installed for a new > customer. > > So, I give up. My new approach is ?VSPs cannot deliver consistent highly > reliable service ? work around it!? > > I know I want a dial plan that will check the bridge status and redial using a > different gateway (VSP) if it failed (not busy or no answer ? failure). I am > pretty sure the best/easy approach is a JavaScript ?dialer? script to handle > this. > > I have looked in the wiki for such a sample. If it exists I have missed it. > So rather than ?re-inventing the wheel? I am reaching out to you all. Does > anyone have a dialing script that checks for failure and redials on failure or > just continues to voicemail on no answer? I will happily test and tweak it > and add it to the wiki if you have something to get me started!!! If not I > shall take the plunge and get started! My ideal solution will dial, play a > message and redial on failure, maybe 2 or three times or play a failure > message and email me on excess errors. I am leaning toward ?Call congestion, > rerouting. Please stand by? while the new bridge is being attempted. > > Thanks, > Sean > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130113/2893eee4/attachment.html From bdfoster at endigotech.com Mon Jan 14 07:55:30 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 13 Jan 2013 23:55:30 -0500 Subject: [Freeswitch-users] Redundant Dial Plan in JS In-Reply-To: <0b1701cdf206$6a567930$3f036b90$@bizfocused.com> References: <0b1701cdf206$6a567930$3f036b90$@bizfocused.com> Message-ID: <985D292F-AEB6-4C2A-9284-006C31F78634@endigotech.com> This is something you can do in standard dialplan xml. See continue_on_fail. Also check out mod_lcr for doing least cost routing and backing it up with another option if one fails. Sent from my iPhone On Jan 13, 2013, at 10:22 PM, "Sean Devoy" wrote: > Hi, > > This question stems from the fact that I cannot find a VOIP Service Provider (VSP) that can connect outgoing calls truly reliably. I have customers who expect calls to go through just like good old ma-bell did ?100% of the time barring natural disasters. You have to admit those guys set the bar pretty high for us. > > I was using Voip Innovations with decent success, but my users complain every 2 weeks or so and I send the trace to VI. The standard response is always ?We have updated our route tables for that area.? I switched to Vitelity where I was assured they have redundant paths and this could never happen! I have only confirmed it happening twice in the last 3 weeks! Of course one of those was during the demo of the new service I had just installed for a new customer. > > So, I give up. My new approach is ?VSPs cannot deliver consistent highly reliable service ? work around it!? > > I know I want a dial plan that will check the bridge status and redial using a different gateway (VSP) if it failed (not busy or no answer ? failure). I am pretty sure the best/easy approach is a JavaScript ?dialer? script to handle this. > > I have looked in the wiki for such a sample. If it exists I have missed it. So rather than ?re-inventing the wheel? I am reaching out to you all. Does anyone have a dialing script that checks for failure and redials on failure or just continues to voicemail on no answer? I will happily test and tweak it and add it to the wiki if you have something to get me started!!! If not I shall take the plunge and get started! My ideal solution will dial, play a message and redial on failure, maybe 2 or three times or play a failure message and email me on excess errors. I am leaning toward ?Call congestion, rerouting. Please stand by? while the new bridge is being attempted. > > Thanks, > Sean > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130113/9b651044/attachment-0001.html From gassaad at emassembly.com Mon Jan 14 07:20:55 2013 From: gassaad at emassembly.com (George Assaad) Date: Sun, 13 Jan 2013 23:20:55 -0500 Subject: [Freeswitch-users] Redundant Dial Plan in JS In-Reply-To: <0b1701cdf206$6a567930$3f036b90$@bizfocused.com> References: <0b1701cdf206$6a567930$3f036b90$@bizfocused.com> Message-ID: You can have continue on fail before bridge to gateway (VSP) ; it will try one after the other until the end of choices. Cheers, George A On Jan 13, 2013 10:28 PM, "Sean Devoy" wrote: > Hi,**** > > ** ** > > This question stems from the fact that I cannot find a VOIP Service > Provider (VSP) that can connect outgoing calls truly reliably. I have > customers who expect calls to go through just like good old ma-bell did > ?100% of the time barring natural disasters. You have to admit those guys > set the bar pretty high for us.**** > > ** ** > > I was using Voip Innovations with decent success, but my users complain > every 2 weeks or so and I send the trace to VI. The standard response is > always ?We have updated our route tables for that area.? I switched to > Vitelity where I was assured they have redundant paths and this could never > happen! I have only confirmed it happening twice in the last 3 weeks! Of > course one of those was during the demo of the new service I had just > installed for a new customer. **** > > ** ** > > So, I give up. My new approach is ?VSPs cannot deliver consistent highly > reliable service ? work around it!?**** > > ** ** > > I know I want a dial plan that will check the bridge status and redial > using a different gateway (VSP) if it failed (not busy or no answer ? > failure). I am pretty sure the best/easy approach is a JavaScript ?dialer? > script to handle this.**** > > ** ** > > I have looked in the wiki for such a sample. If it exists I have missed > it. So rather than ?re-inventing the wheel? I am reaching out to you > all. Does anyone have a dialing script that checks for failure and redials > on failure or just continues to voicemail on no answer? I will happily > test and tweak it and add it to the wiki if you have something to get me > started!!! If not I shall take the plunge and get started! My ideal > solution will dial, play a message and redial on failure, maybe 2 or three > times or play a failure message and email me on excess errors. I am > leaning toward ?Call congestion, rerouting. Please stand by? while the > new bridge is being attempted.**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130113/3500bff4/attachment.html From jtbock at synacktics.com Mon Jan 14 07:39:10 2013 From: jtbock at synacktics.com (Tim Bock) Date: Sun, 13 Jan 2013 21:39:10 -0700 Subject: [Freeswitch-users] dtmf problem Message-ID: <50F38BEE.6090804@synacktics.com> Hello List, I have a freeswitch server with a digium tdm wildcard. Compiled, enabled, and configured freetdm and dahdi...all seems to be working fine (can make/receive calls). My problem is with dtmf. When calling from an external pstn number to freeswitch, dtmf seems to be detected properly and consistently. But when calling from a cell phone or via another voip line, dtmf doesn't seem to be detected, or detected only occasionally. On the same system, if I call into the voip number, calls from any of the three external options (landline, cell, sip) all seem to work fine with dtmf. So the problem seems to be something I've misconfigured with the tdm card. Can anyone point me in the right direction? Thank you, Tim From tarkhil at over.ru Mon Jan 14 08:27:36 2013 From: tarkhil at over.ru (Alex Povolotsky) Date: Mon, 14 Jan 2013 09:27:36 +0400 Subject: [Freeswitch-users] SIP/TLS, INVITE and NAT Message-ID: <50F39748.7010706@over.ru> Hello, I'm trying to set up TLS-protected FS, and cannot resolve TLS+NAT incoming calls issue. My clients are CSipSimple on Android and XLite on PC. Everything works fine when calling to non-TLS client, activating TLS yields (CSipSimple -> XLite): 2013-01-14 05:08:46.941041 [DEBUG] sofia_glue.c:3648 sofia/internal/1001 at sip.over.ru Set rtp dtmf delay to 40 2013-01-14 05:08:46.941041 [NOTICE] sofia_glue.c:4259 Pre-Answer sofia/internal/1001 at sip.over.ru! 2013-01-14 05:08:46.941041 [DEBUG] switch_channel.c:3136 (sofia/internal/1001 at sip.over.ru) Callstate Change RINGING -> EARLY 2013-01-14 05:08:46.941041 [DEBUG] mod_sofia.c:856 Local SDP sofia/internal/1001 at sip.over.ru: v=0 o=FreeSWITCH 1358119830 1358119831 IN IP4 88.85.71.230 s=FreeSWITCH c=IN IP4 88.85.71.230 t=0 0 m=audio 20296 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2013-01-14 05:08:46.941041 [DEBUG] switch_core_session.c:830 Send signal sofia/internal/1001 at sip.over.ru [BREAK] 2013-01-14 05:08:46.941041 [DEBUG] switch_channel.c:3395 (sofia/internal/1001 at sip.over.ru) Callstate Change EARLY -> ACTIVE 2013-01-14 05:08:46.941041 [NOTICE] mod_dptools.c:1176 Channel [sofia/internal/1001 at sip.over.ru] has been answered EXECUTE sofia/internal/1001 at sip.over.ru sleep(1000) 2013-01-14 05:08:46.941041 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001 at sip.over.ru [BREAK] 2013-01-14 05:08:46.941041 [DEBUG] sofia.c:5599 Channel sofia/internal/1001 at sip.over.ru entering state [completed][200] 2013-01-14 05:08:46.981041 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=1379485558 seq=19624 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.101068 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=1379485558 seq=19625 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.121057 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001 at sip.over.ru [BREAK] 2013-01-14 05:08:47.121057 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001 at sip.over.ru [BREAK] 2013-01-14 05:08:47.121057 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/1001 at sip.over.ru [BREAK] 2013-01-14 05:08:47.141041 [DEBUG] sofia.c:5599 Channel sofia/internal/1001 at sip.over.ru entering state [ready][200] 2013-01-14 05:08:47.281041 [INFO] switch_rtp.c:3642 Auto Changing port from 95.27.78.36:8081 to 95.27.78.36:42659 2013-01-14 05:08:47.301066 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=1379485558 seq=19626 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.501043 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=1379485558 seq=19627 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.701074 [DEBUG] switch_rtp.c:928 [ zrtp engine]: WARNING! HELLO have been resent 5 times without a response. Raising ZRTP_EVENT_NO_ZRTP_QUICK event. ID=48 2013-01-14 05:08:47.701074 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=1379485558 seq=19628 size=140. Stream 48:CLEAR:START 2013-01-14 05:08:47.901047 [DEBUG] switch_rtp.c:928 [ zrtp utils]: Send ssrc=1379485558 seq=19629 size=140. Stream 48:CLEAR:START EXECUTE sofia/internal/1001 at sip.over.ru bridge(loopback/app=voicemail:default 88.85.71.230 1000) Calling from XLite to CSIpSimple is somehow different, after about a minute of conversation it breaks, but with the same "HELLO have been resent 5 times" in logs. It is possible at all to interconnect two NATed clients using SIP/TLS? If yes, what softphones proved working and how should I setup FS/phones? Alex From wangd at alongtechnology.com.cn Mon Jan 14 10:47:33 2013 From: wangd at alongtechnology.com.cn (=?utf-8?B?546L6L+q?=) Date: Mon, 14 Jan 2013 15:47:33 +0800 Subject: [Freeswitch-users] =?utf-8?q?How_to_pause_and_resume_play_an_audi?= =?utf-8?q?o_file=09in_conference=3F?= Message-ID: <201301141547320568849@alongtechnology.com.cn> "uuid_fileman" is for channel, not for a conference. If I used "uuid_fileman" to pause a channel, and I can not talk to this channel. I wanna play file in a conference, and pause the playing and talk, then stop talking and resume the playing. ???? freeswitch-users-request ????? 2013-01-10 22:12:37 ???? freeswitch-users ??? ??? FreeSWITCH-users Digest, Vol 79, Issue 54 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: How to pause and resume play an audio file in conference? (Avi Marcus) 2. Re: Best practices question about SIP registration (Steven Ayre) 3. Re: mod_com_g729 transcoding (Steven Ayre) 4. Early media without bridge (Tamas.Cseke ) 5. Loopback Endpoint (Jon Sch?pzinsky) 6. Re: mod_directory menu-top? (Abaci) ------------------------------------------------------------ From: Avi Marcus To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to pause and resume play an audio filein conference? Date: Thu10 Jan 2013 11:31:49 +0200 It looks like you can do it via api command: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman It seems the conference has stop-talking and start-talking events, so you can start the file and then have an ESL app that listens for the events and stops/starts the playback. Let us know how that goes. -Avi On Thu, Jan 10, 2013 at 3:39 AM, ?? wrote: > ** > How to pause and resume play an audio file in conference? For example: > I create a conference room with some members, and play an audio file to > them. When someone start-talking, I pause the playing. When nobody talking, > I resume play the audio file. > > BTW, In freeswitch-1.2rc, I tryed play,pause,resume commands, but pause > and resume is for recording, not for my needs. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------------------------------------ ------------------------------------------------------------ From: Steven Ayre To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best practices question about SIP registration Date: Thu10 Jan 2013 09:31:29 +0000 > There IS a wrinkle to this special case: If you have a URL of the form the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host The reason being that a SRV record includes the port (note that means it's a very easy way to add/change additional ports as well as addresses without having to change customer configs). So obviously a host:port isn't compatible with a SRV, since it wouldn't make sense to override the port, so the assumption is that it's specifying an A/AAAA record instead and port on that IP instead. -Steve On 8 January 2013 23:35, Lawrence Conroy wrote: > Hi Michael, folks, > if you're going to codify it, I did slightly simplify things: > (full disclosure -- I disagreed with it at the time which is probably why I forgot to mention it -- honest). > > There IS a wrinkle to this special case: If you have a URL of the form the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host example.com (i.e., there's a machine called example.com, and it is handling SIP traffic for that domainpart). > Basically, if you see a colon in the domainpart, you're looking for a machine -- otherwise you're looking for a NAPTR (and/or a SRV at _sip._udp.). > > I'd put that before the paragraph starting "However, relying on ..." > > Curiously enough, the old 2543-compliant servers did hunt the SRV rather than giving up and looking for an A, so this was a change. Such fun was had re-writing implementations (plural) and testing them yet again. Sigh. > > all the best, > Lawrence > > > On 8 Jan 2013, at 22:56, Michael Collins wrote: > >> Lawrence, >> >> Thanks for this explanation. It was very well written. I'm looking for a >> place to codify this on the wiki so that it gets preserved... :) >> >> -MC >> >> On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy wrote: >> >>> Hi there, >>> at the risk of butting in on someone else's party ... >>> Nope; your interpretations is NOT best practice. >>> I have some sympathy, as the term domain is overloaded within fS. >>> >>> A sip address consists of a userpart and a domain part -- e.g., >>> >>> The sip domain is similar to an email domain -- e.g., >> user at maildomain> >>> With email, you need to do a lookup of the MX record in DNS to find the >>> FQDN of the machine that handles mail for the domain. >>> With SIP (see RFC 3263), you do a lookup on the SRV record (at >>> _sip._udp.) to find the machine that handles SIP >>> registrations/incalls for the domain. That also gives you the port on >>> which that machine is listening. >>> (Yup, you can also have a NAPTR record in the domain to tell you where the >>> SRV record is, but many folks don't bother -- for Best Practice, you >>> should, but ...) >>> >>> There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) >>> support that means most SIP clients will look for the SRV and, if it can't >>> be found (or there's an IP address rather than a DNS -style domain, in >>> which case the SIP client won't bother hunting the SRV), the client will >>> guess that the domain has a machine (i.e. it will look for an A or AAAA >>> record), and also guess it's listening on 5060 (the default port). >>> Email is the same (mail to fred at example.com, and strictly the sender will >>> do a check for a MX and then look for an A record for example.com, and >>> try there). >>> >>> However, relying on that default "get out" clause is definitely NOT what >>> you should do for BCP. >>> Using the hostname as the sip domain is a kludge -- the FQDN with A record >>> usually works, but it's not what you want to do. >>> >>> SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at >>> _sip._udp., and you're done. No need for an A record at that domain >>> at all. >>> >>> (RFC 3263 is not too hard to read, for a change -- it's certainly shorter >>> than RFC 3261, and it even has an ASCII art diagram :). >>> >>> all the best, >>> Lawrence >>> >>> On 8 Jan 2013, at 21:05, Steven Schoch wrote: >>> >>>> On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < >>>> fs-list at communicatefreely.net> wrote: >>>> >>>>> Hi Steven, >>>>> >>>>> I would recommend using a proper domain name as much as possible. For >>>>> one, it looks >>>>> nicer! A SIP URI is supposed to be user at domain like an e-mail address >>>>> is, and I hope that >>>>> one day URI dialing will be common place, so we might as well do it >>> right >>>>> the first time. >>>>> >>>> >>>> What you're saying is that "domain" should really be a fully-qualified >>> host >>>> name that points via DNS to the actual host on which FreeSwitch is >>> running. >>>> That is, the domain should be "pbx.example.com" instead of just " >>>> example.com", as the last example would most likely point to a web >>> server, >>>> not the SIP server. Do I have that right? >>>> >>>> Next, in the configuration for Polycom phones (for example), there are 2 >>>> fields that both have the userid. In the example in >>>> http://wiki.freeswitch.org/wiki/Polycom_configuration it has: >>>> >>>> reg.1.auth.userId="1000" >>>> >>>> and >>>> >>>> reg.1.address="1000 at fs.domain.local" >>>> >>>> How is the "address" value used? Is that sent in the SIP registration >>>> message? If that's the case, what does Freeswitch do with it? >>>> >>>> -- >>>> Steve >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ------------------------------------------------------------ ------------------------------------------------------------ From: Steven Ayre To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_com_g729 transcoding Date: Thu10 Jan 2013 09:38:52 +0000 A more complete debug-level log could be useful to get more context. Is box 2 running any other calls at the same time that might be using the license? Is box 2 doing anything else with the call? Anything like eavesdrop, recording, start_dtmf / start_dtmf_generate etc which uses the media will use a license, and I *think* that license only gets released until the end of the call. -Steve On 10 January 2013 07:09, Colin Mason wrote: > I have phone A connected to freeswitch box 1 and phone B connected to > freeswitch box 2. > > > > Phone A wants to dial phone B using G729. Codec is always G729. > > The path RTP follows is: > > > > > > > > Phone A -------> FreeSWITCH 1 -------> FreeSWITCH 2 -------> Phone B > > (g729) (g729) > (g729) > > > > > > > > My question is, why is it that the freeswitch box receiving the call uses up > an encoder/decoder when the codec is G729 along the path? If I reverse the > call and call Phone A from Phone B, FreeSWITCH Box 1 uses up 1 license. > > if I dial the PSTN to a carrier who supports G729, I don?t use up a license. > Any thoughts? Maybe this is normal. > > > > FreeSWITCH Box 1: > > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP > [sofia/mpls/I888_1 at 172.17.17.17] 172.17.17.17 port 32014 -> 10.253.200.6 > port 16466 codec: 18 ms: 20 > > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP > [sofia/transport/2996] 172.17.17.17 port 27276 -> 172.16.16.16 port 17056 > codec: 18 ms: 20 > > freeswitch at internal> g729_info > > Permitted G729 channels: 40 > > Encoders in use: 0 > > Decoders in use: 0 > > > > FreeSWITCH Box 2: > > 2013-01-10 01:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP > [sofia/transport/3888 at 172.17.17.17] 172.16.16.16 port 17056 -> 172.17.17.17 > port 27276 codec: 18 ms: 20 > > 2013-01-10 01:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP > [sofia/mpls/sip:C996_1 at 10.253.200.10:5060] 172.16.16.16 port 18966 -> > 10.253.200.10 port 16486 codec: 18 ms: 20 > > freeswitch at internal> g729_info > > Permitted G729 channels: 40 > > Encoders in use: 1 > > Decoders in use: 1 > > > > > > > > Thanks in advance guys. > > Colin > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ------------------------------------------------------------ ------------------------------------------------------------ From: "Tamas.Cseke " To: FreeSWITCH Users Help Subject: [Freeswitch-users] Early media without bridge Date: Thu10 Jan 2013 12:24:00 +0100 Hello, We would like to hear early media without CHANNEL_BRIDGE event These are failed calls and callers would like to hear the message that the provider plays Because the caller thinks the call is answered if originate returns. as far as I understand: -early media makes the originate return -if we ignore early media the bridge won't return, but we don't hear it we would like both of them, is it possible somehow? I 'm not sure I fully understand all of the ignore_early_media options but I haven't find solution for this, Could you please advise me one, if there is any? I'm thinking about we maybe need a new ignore_early_media option like "consume" but sending the media to the caller instead of dropping it If there isn't already a solution I also would appreciate if you let me know your opinion about this idea Thanks advance, Tamas ------------------------------------------------------------ ------------------------------------------------------------ From: Jon_Sch?zinsky To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] Loopback Endpoint Date: Thu10 Jan 2013 14:36:06 +0100 Hello List, I can see that the "this will destroy the world and this may kill your pets" warning has been removed from the documentation for the Loopback endpoint. Is this an indication that it has become safer to use? I have a specific problem where I essentially have to dial a dialplan for each user I am trying to reach, in parallel. Is it correctly understood that loopback would be the best/only way of implementing this, or is there another way? Kind Regards Jon Sch?pzinsky ------------------------------------------------------------ ------------------------------------------------------------ From: Abaci To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_directory menu-top? Date: Thu10 Jan 2013 09:11:09 -0500 use 'execute_extension' to start the directory application so that you get back to the ivr when you exit the directory application. On 1/9/2013 6:41 PM, Phillip Warner wrote: > Hi, is there a parameter in mod_directory to have it transfer (back) to an ivr if the user decides not to search by directory instead of having to hang-up and call again? > > For example: ivr --> directory --> user changes mind about dialling by name and wants to return to ivr --> ivr > > Thanks. Phil. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/9abb8577/attachment-0001.html From itispip-qq at hotmail.com Mon Jan 14 14:53:55 2013 From: itispip-qq at hotmail.com (Pip Live) Date: Mon, 14 Jan 2013 19:53:55 +0800 Subject: [Freeswitch-users] Chatplan: is msg received success? Message-ID: Hi, I use Freeswitch to handle SIP message between SIP extentions, how can server know does the message be received by client or not? I noticed even if a SIP extention is offline/unregister, SIP message will still be sent by server but just the receiver will never get it (even if later the receiver online/registered again). What I want to achieve is: send SIP message to an extention, if received OK, nothing happen; if not, forward the message to user's email address. brgds, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/e2eff6f7/attachment.html From qasimakhan at gmail.com Mon Jan 14 15:04:24 2013 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Mon, 14 Jan 2013 17:04:24 +0500 Subject: [Freeswitch-users] Fwd: In-Reply-To: <50f36b09.232b700a.36b0.4603@mx.google.com> References: <50f36b09.232b700a.36b0.4603@mx.google.com> Message-ID: Is this spam? On Mon, Jan 14, 2013 at 7:17 AM, Woody Dickson wrote: > http://95.110.242.215/6bmhxm.php > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/bb3500b0/attachment.html From andrew at cassidywebservices.co.uk Mon Jan 14 15:35:12 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 14 Jan 2013 12:35:12 +0000 Subject: [Freeswitch-users] Fwd: In-Reply-To: References: <50f36b09.232b700a.36b0.4603@mx.google.com> Message-ID: looks like it to me. On 14 January 2013 12:04, qasimakhan at gmail.com wrote: > Is this spam? > > > On Mon, Jan 14, 2013 at 7:17 AM, Woody Dickson wrote: > >> http://95.110.242.215/6bmhxm.php >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/bddc5a7c/attachment.html From steveayre at gmail.com Mon Jan 14 15:49:33 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 14 Jan 2013 12:49:33 +0000 Subject: [Freeswitch-users] How to pause and resume play an audio file in conference? In-Reply-To: <201301141547320568849@alongtechnology.com.cn> References: <201301141547320568849@alongtechnology.com.cn> Message-ID: Try: conference play conference pause conference resume conference stop http://wiki.freeswitch.org/wiki/Mod_conference On 14 January 2013 07:47, ?? wrote: > ** > "uuid_fileman" is for channel, not for a conference. If I used > "uuid_fileman" to pause a channel, and I can not talk to this channel. I > wanna play file in a conference, and pause the playing and talk, then stop > talking and resume the playing. > > ------------------------------ > *????* freeswitch-users-request > *?????* 2013-01-10 22:12:37 > *????* freeswitch-users > *???* > *???* FreeSWITCH-users Digest, Vol 79, Issue 54 > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > 1. Re: How to pause and resume play an audio file in conference? > (Avi Marcus) > 2. Re: Best practices question about SIP registration (Steven Ayre) > 3. Re: mod_com_g729 transcoding (Steven Ayre) > 4. Early media without bridge (Tamas.Cseke ) > 5. Loopback Endpoint (Jon Sch?pzinsky) > 6. Re: mod_directory menu-top? (Abaci) > ------------------------------------------------------------ > From: Avi Marcus > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] How to pause and resume play an audio filein conference? > Date: Thu10 Jan 2013 11:31:49 +0200 > It looks like you can do it via api command: > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman > It seems the conference has stop-talking and start-talking events, so you > can start the file and then have an ESL app that listens for the events and > stops/starts the playback. > Let us know how that goes. > -Avi > On Thu, Jan 10, 2013 at 3:39 AM, ?? wrote: > > ** > > How to pause and resume play an audio file in conference? For example: > > I create a conference room with some members, and play an audio file to > > > them. When someone start-talking, I pause the playing. When nobody talking, > > I resume play the audio file. > > > > BTW, In freeswitch-1.2rc, I tryed play,pause,resume commands, but pause > > and resume is for recording, not for my needs. > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ------------------------------------------------------------ > ------------------------------------------------------------ > From: Steven Ayre > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Best practices question about SIP registration > Date: Thu10 Jan 2013 09:31:29 +0000 > > > There IS a wrinkle to this special case: If you have a URL of the form < > sip:user at example.com:5055 > > the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host > The reason being that a SRV record includes the port (note that means > it's a very easy way to add/change additional ports as well as > addresses without having to change customer configs). > So obviously a host:port isn't compatible with a SRV, since it > wouldn't make sense to override the port, so the assumption is that > it's specifying an A/AAAA record instead and port on that IP instead. > -Steve > On 8 January 2013 23:35, Lawrence Conroy > wrote: > > Hi Michael, folks, > > if you're going to codify it, I did slightly simplify things: > > > (full disclosure -- I disagreed with it at the time which is probably why I forgot to mention it -- honest). > > > > There IS a wrinkle to this special case: If you have a URL of the form < > sip:user at example.com:5055 > > the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host > example.com (i.e., there's a machine called example.com > , and it is handling SIP traffic for that domainpart). > > > Basically, if you see a colon in the domainpart, you're looking for a machine -- otherwise you're looking for a NAPTR (and/or a SRV at _sip._udp.). > > > > I'd put that before the paragraph starting "However, relying on ..." > > > > > Curiously enough, the old 2543-compliant servers did hunt the SRV rather than giving up and looking for an A, so this was a change. Such fun was had re-writing implementations (plural) and testing them yet again. Sigh. > > > > all the best, > > Lawrence > > > > > > On 8 Jan 2013, at 22:56, Michael Collins wrote: > > > >> Lawrence, > >> > >> Thanks for this explanation. It was very well written. I'm looking for a > >> place to codify this on the wiki so that it gets preserved... :) > >> > >> -MC > >> > >> On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy < > lconroy at insensate.co.uk>wrote: > >> > >>> Hi there, > >>> at the risk of butting in on someone else's party ... > >>> Nope; your interpretations is NOT best practice. > >>> I have some sympathy, as the term domain is overloaded within fS. > >>> > >>> A sip address consists of a userpart and a domain part -- e.g., > >>> > >>> The sip domain is similar to an email domain -- e.g., >>> user at maildomain> > >>> With email, you need to do a lookup of the MX record in DNS to find the > >>> FQDN of the machine that handles mail for the domain. > >>> With SIP (see RFC 3263), you do a lookup on the SRV record (at > >>> _sip._udp.) to find the machine that handles SIP > >>> registrations/incalls for the domain. That also gives you the port on > >>> which that machine is listening. > > >>> (Yup, you can also have a NAPTR record in the domain to tell you where the > >>> SRV record is, but many folks don't bother -- for Best Practice, you > >>> should, but ...) > >>> > >>> There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) > > >>> support that means most SIP clients will look for the SRV and, if it can't > >>> be found (or there's an IP address rather than a DNS -style domain, in > > >>> which case the SIP client won't bother hunting the SRV), the client will > >>> guess that the domain has a machine (i.e. it will look for an A or AAAA > >>> record), and also guess it's listening on 5060 (the default port). > >>> Email is the same (mail to fred at example.com > , and strictly the sender will > >>> do a check for a MX and then look for an A record for example.com, and > >>> try there). > >>> > > >>> However, relying on that default "get out" clause is definitely NOT what > >>> you should do for BCP. > > >>> Using the hostname as the sip domain is a kludge -- the FQDN with A record > >>> usually works, but it's not what you want to do. > >>> > > >>> SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at > > >>> _sip._udp., and you're done. No need for an A record at that domain > >>> at all. > >>> > > >>> (RFC 3263 is not too hard to read, for a change -- it's certainly shorter > >>> than RFC 3261, and it even has an ASCII art diagram :). > >>> > >>> all the best, > >>> Lawrence > >>> > >>> On 8 Jan 2013, at 21:05, Steven Schoch wrote: > >>> > >>>> On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < > >>>> fs-list at communicatefreely.net> wrote: > >>>> > >>>>> Hi Steven, > >>>>> > > >>>>> I would recommend using a proper domain name as much as possible. For > >>>>> one, it looks > >>>>> nicer! A SIP URI is supposed to be user at domain > like an e-mail address > >>>>> is, and I hope that > >>>>> one day URI dialing will be common place, so we might as well do it > >>> right > >>>>> the first time. > >>>>> > >>>> > >>>> What you're saying is that "domain" should really be a fully-qualified > >>> host > >>>> name that points via DNS to the actual host on which FreeSwitch is > >>> running. > >>>> That is, the domain should be "pbx.example.com" instead of just " > >>>> example.com", as the last example would most likely point to a web > >>> server, > >>>> not the SIP server. Do I have that right? > >>>> > > >>>> Next, in the configuration for Polycom phones (for example), there are 2 > >>>> fields that both have the userid. In the example in > >>>> http://wiki.freeswitch.org/wiki/Polycom_configuration it has: > >>>> > >>>> reg.1.auth.userId="1000" > >>>> > >>>> and > >>>> > >>>> reg.1.address="1000 at fs.domain.local" > >>>> > >>>> How is the "address" value used? Is that sent in the SIP registration > >>>> message? If that's the case, what does Freeswitch do with it? > >>>> > >>>> -- > >>>> Steve > > >>>> _________________________________________________________________________ > >>>> Professional FreeSWITCH Consulting Services: > >>>> consulting at freeswitch.org > >>>> http://www.freeswitchsolutions.com > >>>> > >>>> > >>>> > >>>> > >>>> Official FreeSWITCH Sites > >>>> http://www.freeswitch.org > >>>> http://wiki.freeswitch.org > >>>> http://www.cluecon.com > >>>> > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>> > >>> > > >>> _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Michael S Collins > >> Twitter: @mercutioviz > >> http://www.FreeSWITCH.org > >> http://www.ClueCon.com > >> http://www.OSTAG.org > > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > ------------------------------------------------------------ > ------------------------------------------------------------ > From: Steven Ayre > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_com_g729 transcoding > Date: Thu10 Jan 2013 09:38:52 +0000 > A more complete debug-level log could be useful to get more context. > Is box 2 running any other calls at the same time that might be using > the license? > Is box 2 doing anything else with the call? Anything like eavesdrop, > recording, start_dtmf / start_dtmf_generate etc which uses the media > will use a license, and I *think* that license only gets released > until the end of the call. > -Steve > On 10 January 2013 07:09, Colin Mason > wrote: > > I have phone A connected to freeswitch box 1 and phone B connected to > > freeswitch box 2. > > > > > > > > Phone A wants to dial phone B using G729. Codec is always G729. > > > > The path RTP follows is: > > > > > > > > > > > > > > > > Phone A -------> FreeSWITCH 1 -------> FreeSWITCH 2 -------> Phone B > > > > (g729) (g729) > > (g729) > > > > > > > > > > > > > > > > > My question is, why is it that the freeswitch box receiving the call uses up > > > an encoder/decoder when the codec is G729 along the path? If I reverse the > > call and call Phone A from Phone B, FreeSWITCH Box 1 uses up 1 license. > > > > > if I dial the PSTN to a carrier who supports G729, I don?t use up a license. > > Any thoughts? Maybe this is normal. > > > > > > > > FreeSWITCH Box 1: > > > > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP > > [sofia/mpls/I888_1 at 172.17.17.17] 172.17.17.17 port 32014 -> 10.253.200.6 > > port 16466 codec: 18 ms: 20 > > > > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP > > [sofia/transport/2996] 172.17.17.17 port 27276 -> 172.16.16.16 port 17056 > > codec: 18 ms: 20 > > > > freeswitch at internal> g729_info > > > > Permitted G729 channels: 40 > > > > Encoders in use: 0 > > > > Decoders in use: 0 > > > > > > > > FreeSWITCH Box 2: > > > > 2013-01-10 01:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP > > [sofia/transport/3888 at 172.17.17.17 > ] 172.16.16.16 port 17056 -> 172.17.17.17 > > port 27276 codec: 18 ms: 20 > > > > 2013-01-10 01:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP > > [sofia/mpls/sip:C996_1 at 10.253.200.10:5060] 172.16.16.16 port 18966 -> > > 10.253.200.10 port 16486 codec: 18 ms: 20 > > > > freeswitch at internal> g729_info > > > > Permitted G729 channels: 40 > > > > Encoders in use: 1 > > > > Decoders in use: 1 > > > > > > > > > > > > > > > > Thanks in advance guys. > > > > Colin > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > ------------------------------------------------------------ > ------------------------------------------------------------ > From: "Tamas.Cseke " > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Early media without bridge > Date: Thu10 Jan 2013 12:24:00 +0100 > Hello, > We would like to hear early media without CHANNEL_BRIDGE event > These are failed calls and callers would like to hear the message that > the provider plays > Because the caller thinks the call is answered if originate returns. > as far as I understand: > -early media makes the originate return > -if we ignore early media the bridge won't return, but we don't hear it > we would like both of them, is it possible somehow? > I 'm not sure I fully understand all of the ignore_early_media options > but I haven't find solution for this, > Could you please advise me one, if there is any? > I'm thinking about we maybe need a new ignore_early_media option > like "consume" but sending the media to the caller instead of dropping it > If there isn't already a solution I also would appreciate if you let me > know your opinion about this idea > Thanks advance, > Tamas > ------------------------------------------------------------ > ------------------------------------------------------------ > From: Jon_Sch?zinsky > To: "freeswitch-users at lists.freeswitch.org"< > freeswitch-users at lists.freeswitch.org> > Subject: [Freeswitch-users] Loopback Endpoint > Date: Thu10 Jan 2013 14:36:06 +0100 > Hello List, > > I can see that the "this will destroy the world and this may kill your pets" warning has been removed from the documentation for the Loopback endpoint. > Is this an indication that it has become safer to use? > > I have a specific problem where I essentially have to dial a dialplan for each user I am trying to reach, in parallel. > > Is it correctly understood that loopback would be the best/only way of implementing this, or is there another way? > Kind Regards > Jon Sch?pzinsky > ------------------------------------------------------------ > ------------------------------------------------------------ > From: Abaci > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] mod_directory menu-top? > Date: Thu10 Jan 2013 09:11:09 -0500 > use 'execute_extension' to start the directory application so that you > get back to the ivr when you exit the directory application. > On 1/9/2013 6:41 PM, Phillip Warner wrote: > > > Hi, is there a parameter in mod_directory to have it transfer (back) to an ivr if the user decides not to search by directory instead of having to hang-up and call again? > > > > > For example: ivr --> directory --> user changes mind about dialling by name and wants to return to ivr --> ivr > > > > Thanks. Phil. > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > ------------------------------------------------------------ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/3c43ca40/attachment-0001.html From steveayre at gmail.com Mon Jan 14 15:50:02 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 14 Jan 2013 12:50:02 +0000 Subject: [Freeswitch-users] How to pause and resume play an audio file in conference? In-Reply-To: References: <201301141547320568849@alongtechnology.com.cn> Message-ID: Nevermind, just reread your original post On 14 January 2013 12:49, Steven Ayre wrote: > Try: > conference play > conference pause > conference resume > conference stop > > http://wiki.freeswitch.org/wiki/Mod_conference > > > > On 14 January 2013 07:47, ?? wrote: > >> ** >> "uuid_fileman" is for channel, not for a conference. If I used >> "uuid_fileman" to pause a channel, and I can not talk to this channel. I >> wanna play file in a conference, and pause the playing and talk, then stop >> talking and resume the playing. >> >> ------------------------------ >> *????* freeswitch-users-request >> *?????* 2013-01-10 22:12:37 >> *????* freeswitch-users >> *???* >> *???* FreeSWITCH-users Digest, Vol 79, Issue 54 >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> Today's Topics: >> 1. Re: How to pause and resume play an audio file in conference? >> (Avi Marcus) >> 2. Re: Best practices question about SIP registration (Steven Ayre) >> 3. Re: mod_com_g729 transcoding (Steven Ayre) >> 4. Early media without bridge (Tamas.Cseke ) >> 5. Loopback Endpoint (Jon Sch?pzinsky) >> 6. Re: mod_directory menu-top? (Abaci) >> ------------------------------------------------------------ >> From: Avi Marcus >> To: FreeSWITCH Users Help >> >> Subject: Re: [Freeswitch-users] How to pause and resume play an audio filein conference? >> Date: Thu10 Jan 2013 11:31:49 +0200 >> It looks like you can do it via api command: >> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman >> >> It seems the conference has stop-talking and start-talking events, so you >> >> can start the file and then have an ESL app that listens for the events and >> stops/starts the playback. >> Let us know how that goes. >> -Avi >> On Thu, Jan 10, 2013 at 3:39 AM, ?? > > wrote: >> > ** >> > How to pause and resume play an audio file in conference? For example: >> > I create a conference room with some members, and play an audio file to >> >> > them. When someone start-talking, I pause the playing. When nobody talking, >> > I resume play the audio file. >> > >> > BTW, In freeswitch-1.2rc, I tryed play,pause,resume commands, but pause >> > and resume is for recording, not for my needs. >> > >> > >> >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> ------------------------------------------------------------ >> ------------------------------------------------------------ >> From: Steven Ayre >> To: FreeSWITCH Users Help >> >> Subject: Re: [Freeswitch-users] Best practices question about SIP registration >> Date: Thu10 Jan 2013 09:31:29 +0000 >> >> > There IS a wrinkle to this special case: If you have a URL of the form < >> sip:user at example.com:5055 >> > the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host >> The reason being that a SRV record includes the port (note that means >> it's a very easy way to add/change additional ports as well as >> addresses without having to change customer configs). >> So obviously a host:port isn't compatible with a SRV, since it >> wouldn't make sense to override the port, so the assumption is that >> it's specifying an A/AAAA record instead and port on that IP instead. >> -Steve >> On 8 January 2013 23:35, Lawrence Conroy > > wrote: >> > Hi Michael, folks, >> > if you're going to codify it, I did slightly simplify things: >> >> > (full disclosure -- I disagreed with it at the time which is probably why I forgot to mention it -- honest). >> > >> > There IS a wrinkle to this special case: If you have a URL of the form < >> sip:user at example.com:5055 >> > the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host >> example.com (i.e., there's a machine called example.com >> , and it is handling SIP traffic for that domainpart). >> >> > Basically, if you see a colon in the domainpart, you're looking for a machine -- otherwise you're looking for a NAPTR (and/or a SRV at _sip._udp.). >> > >> > I'd put that before the paragraph starting "However, relying on ..." >> > >> >> > Curiously enough, the old 2543-compliant servers did hunt the SRV rather than giving up and looking for an A, so this was a change. Such fun was had re-writing implementations (plural) and testing them yet again. Sigh. >> > >> > all the best, >> > Lawrence >> > >> > >> > On 8 Jan 2013, at 22:56, Michael Collins wrote: >> > >> >> Lawrence, >> >> >> >> >> Thanks for this explanation. It was very well written. I'm looking for a >> >> place to codify this on the wiki so that it gets preserved... :) >> >> >> >> -MC >> >> >> >> On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy < >> lconroy at insensate.co.uk>wrote: >> >> >> >>> Hi there, >> >>> at the risk of butting in on someone else's party ... >> >>> Nope; your interpretations is NOT best practice. >> >>> I have some sympathy, as the term domain is overloaded within fS. >> >>> >> >>> A sip address consists of a userpart and a domain part -- e.g., >> >>> >> >>> The sip domain is similar to an email domain -- e.g., > >>> user at maildomain> >> >> >>> With email, you need to do a lookup of the MX record in DNS to find the >> >>> FQDN of the machine that handles mail for the domain. >> >>> With SIP (see RFC 3263), you do a lookup on the SRV record (at >> >>> _sip._udp.) to find the machine that handles SIP >> >>> registrations/incalls for the domain. That also gives you the port on >> >>> which that machine is listening. >> >> >>> (Yup, you can also have a NAPTR record in the domain to tell you where the >> >>> SRV record is, but many folks don't bother -- for Best Practice, you >> >>> should, but ...) >> >>> >> >>> There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) >> >> >>> support that means most SIP clients will look for the SRV and, if it can't >> >>> be found (or there's an IP address rather than a DNS -style domain, in >> >> >>> which case the SIP client won't bother hunting the SRV), the client will >> >> >>> guess that the domain has a machine (i.e. it will look for an A or AAAA >> >>> record), and also guess it's listening on 5060 (the default port). >> >>> Email is the same (mail to fred at example.com >> , and strictly the sender will >> >>> do a check for a MX and then look for an A record for example.com >> , and >> >>> try there). >> >>> >> >> >>> However, relying on that default "get out" clause is definitely NOT what >> >>> you should do for BCP. >> >> >>> Using the hostname as the sip domain is a kludge -- the FQDN with A record >> >>> usually works, but it's not what you want to do. >> >>> >> >> >>> SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at >> >> >>> _sip._udp., and you're done. No need for an A record at that domain >> >>> at all. >> >>> >> >> >>> (RFC 3263 is not too hard to read, for a change -- it's certainly shorter >> >>> than RFC 3261, and it even has an ASCII art diagram :). >> >>> >> >>> all the best, >> >>> Lawrence >> >>> >> >>> On 8 Jan 2013, at 21:05, Steven Schoch wrote: >> >>> >> >>>> On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < >> >>>> fs-list at communicatefreely.net> wrote: >> >>>> >> >>>>> Hi Steven, >> >>>>> >> >> >>>>> I would recommend using a proper domain name as much as possible. For >> >>>>> one, it looks >> >>>>> nicer! A SIP URI is supposed to be user at domain >> like an e-mail address >> >>>>> is, and I hope that >> >>>>> one day URI dialing will be common place, so we might as well do it >> >>> right >> >>>>> the first time. >> >>>>> >> >>>> >> >> >>>> What you're saying is that "domain" should really be a fully-qualified >> >>> host >> >>>> name that points via DNS to the actual host on which FreeSwitch is >> >>> running. >> >>>> That is, the domain should be "pbx.example.com" instead of just " >> >>>> example.com", as the last example would most likely point to a web >> >>> server, >> >>>> not the SIP server. Do I have that right? >> >>>> >> >> >>>> Next, in the configuration for Polycom phones (for example), there are 2 >> >>>> fields that both have the userid. In the example in >> >>>> http://wiki.freeswitch.org/wiki/Polycom_configuration it has: >> >>>> >> >>>> reg.1.auth.userId="1000" >> >>>> >> >>>> and >> >>>> >> >>>> reg.1.address="1000 at fs.domain.local" >> >>>> >> >> >>>> How is the "address" value used? Is that sent in the SIP registration >> >>>> message? If that's the case, what does Freeswitch do with it? >> >>>> >> >>>> -- >> >>>> Steve >> >> >>>> _________________________________________________________________________ >> >>>> Professional FreeSWITCH Consulting Services: >> >>>> consulting at freeswitch.org >> >>>> http://www.freeswitchsolutions.com >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> Official FreeSWITCH Sites >> >>>> http://www.freeswitch.org >> >>>> http://wiki.freeswitch.org >> >>>> http://www.cluecon.com >> >>>> >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>>> http://www.freeswitch.org >> >>> >> >>> >> >> >>> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> >> >> >> >> -- >> >> Michael S Collins >> >> Twitter: @mercutioviz >> >> http://www.FreeSWITCH.org >> >> http://www.ClueCon.com >> >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> ------------------------------------------------------------ >> ------------------------------------------------------------ >> From: Steven Ayre >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] mod_com_g729 transcoding >> Date: Thu10 Jan 2013 09:38:52 +0000 >> A more complete debug-level log could be useful to get more context. >> Is box 2 running any other calls at the same time that might be using >> the license? >> Is box 2 doing anything else with the call? Anything like eavesdrop, >> recording, start_dtmf / start_dtmf_generate etc which uses the media >> will use a license, and I *think* that license only gets released >> until the end of the call. >> -Steve >> On 10 January 2013 07:09, Colin Mason > > wrote: >> > I have phone A connected to freeswitch box 1 and phone B connected to >> > freeswitch box 2. >> > >> > >> > >> > Phone A wants to dial phone B using G729. Codec is always G729. >> > >> > The path RTP follows is: >> > >> > >> > >> > >> > >> > >> > >> > Phone A -------> FreeSWITCH 1 -------> FreeSWITCH 2 -------> Phone B >> > >> > (g729) (g729) >> > (g729) >> > >> > >> > >> > >> > >> > >> > >> >> > My question is, why is it that the freeswitch box receiving the call uses up >> >> > an encoder/decoder when the codec is G729 along the path? If I reverse the >> > call and call Phone A from Phone B, FreeSWITCH Box 1 uses up 1 license. >> > >> >> > if I dial the PSTN to a carrier who supports G729, I don?t use up a license. >> > Any thoughts? Maybe this is normal. >> > >> > >> > >> > FreeSWITCH Box 1: >> > >> > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP >> > [sofia/mpls/I888_1 at 172.17.17.17 >> ] 172.17.17.17 port 32014 -> 10.253.200.6 >> > port 16466 codec: 18 ms: 20 >> > >> > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP >> >> > [sofia/transport/2996] 172.17.17.17 port 27276 -> 172.16.16.16 port 17056 >> > codec: 18 ms: 20 >> > >> > freeswitch at internal> g729_info >> > >> > Permitted G729 channels: 40 >> > >> > Encoders in use: 0 >> > >> > Decoders in use: 0 >> > >> > >> > >> > FreeSWITCH Box 2: >> > >> > 2013-01-10 01:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP >> > [sofia/transport/3888 at 172.17.17.17 >> ] 172.16.16.16 port 17056 -> 172.17.17.17 >> > port 27276 codec: 18 ms: 20 >> > >> > 2013-01-10 01:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP >> > [sofia/mpls/sip:C996_1 at 10.253.200.10:5060] 172.16.16.16 port 18966 -> >> > 10.253.200.10 port 16486 codec: 18 ms: 20 >> > >> > freeswitch at internal> g729_info >> > >> > Permitted G729 channels: 40 >> > >> > Encoders in use: 1 >> > >> > Decoders in use: 1 >> > >> > >> > >> > >> > >> > >> > >> > Thanks in advance guys. >> > >> > Colin >> > >> > >> >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> ------------------------------------------------------------ >> ------------------------------------------------------------ >> From: "Tamas.Cseke " >> To: FreeSWITCH Users Help >> Subject: [Freeswitch-users] Early media without bridge >> Date: Thu10 Jan 2013 12:24:00 +0100 >> Hello, >> We would like to hear early media without CHANNEL_BRIDGE event >> These are failed calls and callers would like to hear the message that >> the provider plays >> Because the caller thinks the call is answered if originate returns. >> as far as I understand: >> -early media makes the originate return >> -if we ignore early media the bridge won't return, but we don't hear it >> we would like both of them, is it possible somehow? >> I 'm not sure I fully understand all of the ignore_early_media options >> but I haven't find solution for this, >> Could you please advise me one, if there is any? >> I'm thinking about we maybe need a new ignore_early_media option >> like "consume" but sending the media to the caller instead of dropping it >> If there isn't already a solution I also would appreciate if you let me >> know your opinion about this idea >> Thanks advance, >> Tamas >> ------------------------------------------------------------ >> ------------------------------------------------------------ >> From: Jon_Sch?zinsky >> To: "freeswitch-users at lists.freeswitch.org"< >> freeswitch-users at lists.freeswitch.org> >> Subject: [Freeswitch-users] Loopback Endpoint >> Date: Thu10 Jan 2013 14:36:06 +0100 >> Hello List, >> >> I can see that the "this will destroy the world and this may kill your pets" warning has been removed from the documentation for the Loopback endpoint. >> Is this an indication that it has become safer to use? >> >> I have a specific problem where I essentially have to dial a dialplan for each user I am trying to reach, in parallel. >> >> Is it correctly understood that loopback would be the best/only way of implementing this, or is there another way? >> Kind Regards >> Jon Sch?pzinsky >> ------------------------------------------------------------ >> ------------------------------------------------------------ >> From: Abaci >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] mod_directory menu-top? >> Date: Thu10 Jan 2013 09:11:09 -0500 >> use 'execute_extension' to start the directory application so that you >> get back to the ivr when you exit the directory application. >> On 1/9/2013 6:41 PM, Phillip Warner wrote: >> >> > Hi, is there a parameter in mod_directory to have it transfer (back) to an ivr if the user decides not to search by directory instead of having to hang-up and call again? >> > >> >> > For example: ivr --> directory --> user changes mind about dialling by name and wants to return to ivr --> ivr >> > >> > Thanks. Phil. >> >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> ------------------------------------------------------------ >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/856881a0/attachment-0001.html From steveayre at gmail.com Mon Jan 14 15:51:14 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 14 Jan 2013 12:51:14 +0000 Subject: [Freeswitch-users] How to pause and resume play an audio file in conference? In-Reply-To: References: <201301141547320568849@alongtechnology.com.cn> Message-ID: I suspect it needs a new feature to be added to mod_conference. You could file a wishlist Jira. Or as a workaround, originate a channel that calls into the conference and play the audio on that channel. You'd then be able to use uuid_fileman on that channel. -Steve On 14 January 2013 12:50, Steven Ayre wrote: > Nevermind, just reread your original post > > > On 14 January 2013 12:49, Steven Ayre wrote: > >> Try: >> conference play >> conference pause >> conference resume >> conference stop >> >> http://wiki.freeswitch.org/wiki/Mod_conference >> >> >> >> On 14 January 2013 07:47, ?? wrote: >> >>> ** >>> "uuid_fileman" is for channel, not for a conference. If I used >>> "uuid_fileman" to pause a channel, and I can not talk to this channel. I >>> wanna play file in a conference, and pause the playing and talk, then stop >>> talking and resume the playing. >>> >>> ------------------------------ >>> *????* freeswitch-users-request >>> *?????* 2013-01-10 22:12:37 >>> *????* freeswitch-users >>> *???* >>> *???* FreeSWITCH-users Digest, Vol 79, Issue 54 >>> Send FreeSWITCH-users mailing list submissions to >>> freeswitch-users at lists.freeswitch.org >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request at lists.freeswitch.org >>> You can reach the person managing the list at >>> freeswitch-users-owner at lists.freeswitch.org >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of FreeSWITCH-users digest..." >>> Today's Topics: >>> 1. Re: How to pause and resume play an audio file in conference? >>> (Avi Marcus) >>> 2. Re: Best practices question about SIP registration (Steven Ayre) >>> 3. Re: mod_com_g729 transcoding (Steven Ayre) >>> 4. Early media without bridge (Tamas.Cseke ) >>> 5. Loopback Endpoint (Jon Sch?pzinsky) >>> 6. Re: mod_directory menu-top? (Abaci) >>> ------------------------------------------------------------ >>> From: Avi Marcus >>> To: FreeSWITCH Users Help >>> >>> Subject: Re: [Freeswitch-users] How to pause and resume play an audio filein conference? >>> Date: Thu10 Jan 2013 11:31:49 +0200 >>> It looks like you can do it via api command: >>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman >>> >>> It seems the conference has stop-talking and start-talking events, so you >>> >>> can start the file and then have an ESL app that listens for the events and >>> stops/starts the playback. >>> Let us know how that goes. >>> -Avi >>> On Thu, Jan 10, 2013 at 3:39 AM, ?? >> > wrote: >>> > ** >>> >>> > How to pause and resume play an audio file in conference? For example: >>> > I create a conference room with some members, and play an audio file to >>> >>> > them. When someone start-talking, I pause the playing. When nobody talking, >>> > I resume play the audio file. >>> > >>> > BTW, In freeswitch-1.2rc, I tryed play,pause,resume commands, but pause >>> > and resume is for recording, not for my needs. >>> > >>> > >>> >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> ------------------------------------------------------------ >>> ------------------------------------------------------------ >>> From: Steven Ayre >>> To: FreeSWITCH Users Help >>> >>> Subject: Re: [Freeswitch-users] Best practices question about SIP registration >>> Date: Thu10 Jan 2013 09:31:29 +0000 >>> >>> > There IS a wrinkle to this special case: If you have a URL of the form < >>> sip:user at example.com:5055 >>> > the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host >>> The reason being that a SRV record includes the port (note that means >>> it's a very easy way to add/change additional ports as well as >>> addresses without having to change customer configs). >>> So obviously a host:port isn't compatible with a SRV, since it >>> wouldn't make sense to override the port, so the assumption is that >>> it's specifying an A/AAAA record instead and port on that IP instead. >>> -Steve >>> On 8 January 2013 23:35, Lawrence Conroy >> > wrote: >>> > Hi Michael, folks, >>> > if you're going to codify it, I did slightly simplify things: >>> >>> > (full disclosure -- I disagreed with it at the time which is probably why I forgot to mention it -- honest). >>> > >>> >>> > There IS a wrinkle to this special case: If you have a URL of the form < >>> sip:user at example.com:5055 >>> > the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host >>> example.com (i.e., there's a machine called example.com >>> , and it is handling SIP traffic for that domainpart). >>> >>> > Basically, if you see a colon in the domainpart, you're looking for a machine -- otherwise you're looking for a NAPTR (and/or a SRV at _sip._udp.). >>> > >>> > I'd put that before the paragraph starting "However, relying on ..." >>> > >>> >>> > Curiously enough, the old 2543-compliant servers did hunt the SRV rather than giving up and looking for an A, so this was a change. Such fun was had re-writing implementations (plural) and testing them yet again. Sigh. >>> > >>> > all the best, >>> > Lawrence >>> > >>> > >>> > On 8 Jan 2013, at 22:56, Michael Collins wrote: >>> > >>> >> Lawrence, >>> >> >>> >>> >> Thanks for this explanation. It was very well written. I'm looking for a >>> >> place to codify this on the wiki so that it gets preserved... :) >>> >> >>> >> -MC >>> >> >>> >> On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy < >>> lconroy at insensate.co.uk>wrote: >>> >> >>> >>> Hi there, >>> >>> at the risk of butting in on someone else's party ... >>> >>> Nope; your interpretations is NOT best practice. >>> >>> I have some sympathy, as the term domain is overloaded within fS. >>> >>> >>> >>> A sip address consists of a userpart and a domain part -- e.g., >>> >>> >>> >>> The sip domain is similar to an email domain -- e.g., >> >>> user at maildomain> >>> >>> >>> With email, you need to do a lookup of the MX record in DNS to find the >>> >>> FQDN of the machine that handles mail for the domain. >>> >>> With SIP (see RFC 3263), you do a lookup on the SRV record (at >>> >>> _sip._udp.) to find the machine that handles SIP >>> >>> registrations/incalls for the domain. That also gives you the port on >>> >>> which that machine is listening. >>> >>> >>> (Yup, you can also have a NAPTR record in the domain to tell you where the >>> >>> SRV record is, but many folks don't bother -- for Best Practice, you >>> >>> should, but ...) >>> >>> >>> >>> >>> There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) >>> >>> >>> support that means most SIP clients will look for the SRV and, if it can't >>> >>> >>> be found (or there's an IP address rather than a DNS -style domain, in >>> >>> >>> which case the SIP client won't bother hunting the SRV), the client will >>> >>> >>> guess that the domain has a machine (i.e. it will look for an A or AAAA >>> >>> record), and also guess it's listening on 5060 (the default port). >>> >>> Email is the same (mail to fred at example.com >>> , and strictly the sender will >>> >>> do a check for a MX and then look for an A record for example.com >>> , and >>> >>> try there). >>> >>> >>> >>> >>> However, relying on that default "get out" clause is definitely NOT what >>> >>> you should do for BCP. >>> >>> >>> Using the hostname as the sip domain is a kludge -- the FQDN with A record >>> >>> usually works, but it's not what you want to do. >>> >>> >>> >>> >>> SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at >>> >>> >>> _sip._udp., and you're done. No need for an A record at that domain >>> >>> at all. >>> >>> >>> >>> >>> (RFC 3263 is not too hard to read, for a change -- it's certainly shorter >>> >>> than RFC 3261, and it even has an ASCII art diagram :). >>> >>> >>> >>> all the best, >>> >>> Lawrence >>> >>> >>> >>> On 8 Jan 2013, at 21:05, Steven Schoch wrote: >>> >>> >>> >>>> On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < >>> >>>> fs-list at communicatefreely.net> wrote: >>> >>>> >>> >>>>> Hi Steven, >>> >>>>> >>> >>> >>>>> I would recommend using a proper domain name as much as possible. For >>> >>>>> one, it looks >>> >>>>> nicer! A SIP URI is supposed to be user at domain >>> like an e-mail address >>> >>>>> is, and I hope that >>> >>>>> one day URI dialing will be common place, so we might as well do it >>> >>> right >>> >>>>> the first time. >>> >>>>> >>> >>>> >>> >>> >>>> What you're saying is that "domain" should really be a fully-qualified >>> >>> host >>> >>>> name that points via DNS to the actual host on which FreeSwitch is >>> >>> running. >>> >>>> That is, the domain should be "pbx.example.com" instead of just " >>> >>>> example.com", as the last example would most likely point to a web >>> >>> server, >>> >>>> not the SIP server. Do I have that right? >>> >>>> >>> >>> >>>> Next, in the configuration for Polycom phones (for example), there are 2 >>> >>>> fields that both have the userid. In the example in >>> >>>> http://wiki.freeswitch.org/wiki/Polycom_configuration it has: >>> >>>> >>> >>>> reg.1.auth.userId="1000" >>> >>>> >>> >>>> and >>> >>>> >>> >>>> reg.1.address="1000 at fs.domain.local" >>> >>>> >>> >>> >>>> How is the "address" value used? Is that sent in the SIP registration >>> >>>> message? If that's the case, what does Freeswitch do with it? >>> >>>> >>> >>>> -- >>> >>>> Steve >>> >>> >>>> _________________________________________________________________________ >>> >>>> Professional FreeSWITCH Consulting Services: >>> >>>> consulting at freeswitch.org >>> >>>> http://www.freeswitchsolutions.com >>> >>>> >>> >>>> >>> >>>> >>> >>>> >>> >>>> Official FreeSWITCH Sites >>> >>>> http://www.freeswitch.org >>> >>>> http://wiki.freeswitch.org >>> >>>> http://www.cluecon.com >>> >>>> >>> >>>> FreeSWITCH-users mailing list >>> >>>> FreeSWITCH-users at lists.freeswitch.org >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>> http://www.freeswitch.org >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> http://www.cluecon.com >>> >>> >>> >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >> >>> >> >>> >> >>> >> -- >>> >> Michael S Collins >>> >> Twitter: @mercutioviz >>> >> http://www.FreeSWITCH.org >>> >> http://www.ClueCon.com >>> >> http://www.OSTAG.org >>> >>> >> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> ------------------------------------------------------------ >>> ------------------------------------------------------------ >>> From: Steven Ayre >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] mod_com_g729 transcoding >>> Date: Thu10 Jan 2013 09:38:52 +0000 >>> A more complete debug-level log could be useful to get more context. >>> Is box 2 running any other calls at the same time that might be using >>> the license? >>> Is box 2 doing anything else with the call? Anything like eavesdrop, >>> recording, start_dtmf / start_dtmf_generate etc which uses the media >>> will use a license, and I *think* that license only gets released >>> until the end of the call. >>> -Steve >>> On 10 January 2013 07:09, Colin Mason >> > wrote: >>> > I have phone A connected to freeswitch box 1 and phone B connected to >>> > freeswitch box 2. >>> > >>> > >>> > >>> > Phone A wants to dial phone B using G729. Codec is always G729. >>> > >>> > The path RTP follows is: >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > Phone A -------> FreeSWITCH 1 -------> FreeSWITCH 2 -------> Phone B >>> > >>> > (g729) (g729) >>> > (g729) >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> >>> > My question is, why is it that the freeswitch box receiving the call uses up >>> >>> > an encoder/decoder when the codec is G729 along the path? If I reverse the >>> > call and call Phone A from Phone B, FreeSWITCH Box 1 uses up 1 license. >>> > >>> >>> > if I dial the PSTN to a carrier who supports G729, I don?t use up a license. >>> > Any thoughts? Maybe this is normal. >>> > >>> > >>> > >>> > FreeSWITCH Box 1: >>> > >>> > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP >>> > [sofia/mpls/I888_1 at 172.17.17.17 >>> ] 172.17.17.17 port 32014 -> 10.253.200.6 >>> > port 16466 codec: 18 ms: 20 >>> > >>> > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP >>> >>> > [sofia/transport/2996] 172.17.17.17 port 27276 -> 172.16.16.16 port 17056 >>> > codec: 18 ms: 20 >>> > >>> > freeswitch at internal> g729_info >>> > >>> > Permitted G729 channels: 40 >>> > >>> > Encoders in use: 0 >>> > >>> > Decoders in use: 0 >>> > >>> > >>> > >>> > FreeSWITCH Box 2: >>> > >>> > 2013-01-10 01:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP >>> > [sofia/transport/3888 at 172.17.17.17 >>> ] 172.16.16.16 port 17056 -> 172.17.17.17 >>> > port 27276 codec: 18 ms: 20 >>> > >>> > 2013-01-10 01:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP >>> > [sofia/mpls/sip:C996_1 at 10.253.200.10:5060] 172.16.16.16 port 18966 -> >>> > 10.253.200.10 port 16486 codec: 18 ms: 20 >>> > >>> > freeswitch at internal> g729_info >>> > >>> > Permitted G729 channels: 40 >>> > >>> > Encoders in use: 1 >>> > >>> > Decoders in use: 1 >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > Thanks in advance guys. >>> > >>> > Colin >>> > >>> > >>> >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> ------------------------------------------------------------ >>> ------------------------------------------------------------ >>> From: "Tamas.Cseke " >>> To: FreeSWITCH Users Help >>> Subject: [Freeswitch-users] Early media without bridge >>> Date: Thu10 Jan 2013 12:24:00 +0100 >>> Hello, >>> We would like to hear early media without CHANNEL_BRIDGE event >>> These are failed calls and callers would like to hear the message that >>> the provider plays >>> Because the caller thinks the call is answered if originate returns. >>> as far as I understand: >>> -early media makes the originate return >>> >>> -if we ignore early media the bridge won't return, but we don't hear it >>> we would like both of them, is it possible somehow? >>> I 'm not sure I fully understand all of the ignore_early_media options >>> but I haven't find solution for this, >>> Could you please advise me one, if there is any? >>> I'm thinking about we maybe need a new ignore_early_media option >>> like "consume" but sending the media to the caller instead of dropping it >>> If there isn't already a solution I also would appreciate if you let me >>> know your opinion about this idea >>> Thanks advance, >>> Tamas >>> ------------------------------------------------------------ >>> ------------------------------------------------------------ >>> From: Jon_Sch?zinsky >>> To: "freeswitch-users at lists.freeswitch.org"< >>> freeswitch-users at lists.freeswitch.org> >>> Subject: [Freeswitch-users] Loopback Endpoint >>> Date: Thu10 Jan 2013 14:36:06 +0100 >>> Hello List, >>> >>> I can see that the "this will destroy the world and this may kill your pets" warning has been removed from the documentation for the Loopback endpoint. >>> Is this an indication that it has become safer to use? >>> >>> I have a specific problem where I essentially have to dial a dialplan for each user I am trying to reach, in parallel. >>> >>> Is it correctly understood that loopback would be the best/only way of implementing this, or is there another way? >>> Kind Regards >>> Jon Sch?pzinsky >>> ------------------------------------------------------------ >>> ------------------------------------------------------------ >>> From: Abaci >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] mod_directory menu-top? >>> Date: Thu10 Jan 2013 09:11:09 -0500 >>> use 'execute_extension' to start the directory application so that you >>> get back to the ivr when you exit the directory application. >>> On 1/9/2013 6:41 PM, Phillip Warner wrote: >>> >>> > Hi, is there a parameter in mod_directory to have it transfer (back) to an ivr if the user decides not to search by directory instead of having to hang-up and call again? >>> > >>> >>> > For example: ivr --> directory --> user changes mind about dialling by name and wants to return to ivr --> ivr >>> > >>> > Thanks. Phil. >>> >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> ------------------------------------------------------------ >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/f96394c2/attachment-0001.html From luis.azedo at factorlusitano.com Mon Jan 14 16:04:05 2013 From: luis.azedo at factorlusitano.com (Luis Azedo) Date: Mon, 14 Jan 2013 13:04:05 +0000 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes Message-ID: Hi, is there any way to enable mod_amr with transcoding for testing purposes ? maybe with opencore-amr ? thanks From mike at jerris.com Mon Jan 14 17:10:17 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 14 Jan 2013 09:10:17 -0500 Subject: [Freeswitch-users] (no subject) In-Reply-To: References: <50f36b09.232b700a.36b0.4603@mx.google.com> Message-ID: sorry about that folks, he'll be removed from the list right now. Mike On Jan 14, 2013, at 7:35 AM, Andrew Cassidy wrote: > looks like it to me. > > On 14 January 2013 12:04, qasimakhan at gmail.com wrote: > Is this spam? > > > On Mon, Jan 14, 2013 at 7:17 AM, Woody Dickson wrote: > http://95.110.242.215/6bmhxm.php > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Andrew Cassidy BSc (Hons) MBCS SSCA > Managing Director > > > T 03300 100 960 F 03300 100 961 > E andrew at cassidywebservices.co.uk > W www.cassidywebservices.co.uk > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/7453a681/attachment.html From royj at yandex.ru Mon Jan 14 17:31:38 2013 From: royj at yandex.ru (royj) Date: Mon, 14 Jan 2013 18:31:38 +0400 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 79, Issue 73 In-Reply-To: References: Message-ID: <20130114183138.8bf7de94ec477cbd8acc423f@yandex.ru> try > > I need to set up a passthrough zrtp in freeswitch. > I followed this wiki: http://wiki.freeswitch.org/wiki/ZRTP and setup two > Jitsi softphones. > > MiTM: works well but with different SAS in the softphones. It is ok because > of a missing zrtp enrollment process. > Passthrough: no SAS at all and Jitsi says the media is not encrypted > I tried to achieve passthrough by uncommenting > sip_profiles/internal.xml: > > > What is the proper way to set up passthrough zrtp and having an end-to-end > zrtp session with the same SAS on both softphones? > > > Thank you, > Peter regards royj From bpriddy at bryantschools.org Mon Jan 14 17:45:32 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Mon, 14 Jan 2013 08:45:32 -0600 Subject: [Freeswitch-users] Delay Round Two...FIGHT! Message-ID: Ok everyone, hope your new years went wonderful! So here is the situation again. THE DELAY IS BACK! I have wiped FS and installed the latest git head and the delay is still present. I want to eliminate all FS options that could fix it before I call AT&T. Any help is appreciated! Person from outside the school calls, secretary answers, secretary hears they person saying hello, secretary says hello over and over again for 10 seconds. Then the B-Leg audio finally is heard by the A-LEG. Note* The B-Leg (secretary) can hear the A-leg talking the whole time. Just the A-Leg cannot hear the (secretary) B-Leg until about 7-13 seconds has passed. I see nothing in the FS log after the audio from the B-Leg finally traverses to the A-Leg. Please help our school fellow FS users! LONG LIVE FS!!! :) -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/849f29c3/attachment.html From miha at softnet.si Mon Jan 14 17:54:29 2013 From: miha at softnet.si (Miha) Date: Mon, 14 Jan 2013 15:54:29 +0100 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: Message-ID: <50F41C25.3020204@softnet.si> Robert, what do you do in your dialplan thatmod_spandsp_fax.c is called and t38 request qets out? I am doing like: {ignore_early_media=true,absolute_codec_string='PCMU,PCMA',fax_enable_t38=true,fax_verbose=true,fax_use_ecm=true,fax_enable_t38_request=true}sofia/gateway/default/... is wiritten on wiki but still no luck. thanks! Miha Dne 1/12/2013 10:05 PM, pis(e Robert Craig: > What did you do on the config side to get it to work? I've tried both > an Ata and a fax to email extension and they both fail in both directions. > > Robert > > On Jan 12, 2013, at 2:04 PM, "George Assaad" > wrote: > >> I use Fusion successfully to send / receive fax. >> The way it works: >> ATA --->FaxServer. >> After successful receipt, Fax Server---> DID where you >> have dialled originally. >> >> Cheers, >> >> George >> >> >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig >> > wrote: >> >> Did anyone see the issue or what I may have missed in the config? >> As mentioned earlier, I thought it would after a fresh install. >> Any help is appreciated. >> >> Robert >> >> >> ------------------------------------------------------------------------ >> From: craig.robert at hotmail.com >> To: freeswitch-users at lists.freeswitch.org >> >> Date: Thu, 10 Jan 2013 13:43:09 -0700 >> >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Thanks for the quick response. Below is the capture from the log >> viewer. In this test, I am using a remote fax machine (line from >> Cox with physical machine) faxing into the "email server" where I >> created an extension for fax-to-email. Calling number is >> 1479-246-0547 . >> >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:569 (sofia/external/+14792460547 >> @flowroute.com ) State >> DESTROY going to sleep >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 >> sofia/external/+14792460547 @flowroute.com >> Standard DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 >> sofia/external/+14792460547 @flowroute.com >> SOFIA DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:569 (sofia/external/+14792460547 >> @flowroute.com ) State >> DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:559 (sofia/external/+14792460547 >> @flowroute.com ) >> Running State Change CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:556 (sofia/external/+14792460547 >> @flowroute.com ) >> Callstate Change HANGUP -> DOWN >> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 >> Close Channel sofia/external/+14792460547 >> @flowroute.com >> [CS_DESTROY] >> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 >> Session 26 (sofia/external/+14792460547 >> @flowroute.com ) Ended >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 >> Session 26 (sofia/external/+14792460547 >> @flowroute.com ) >> Locked, Waiting on external entities >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 >> Send signal sofia/external/+14792460547 >> @flowroute.com [BREAK] >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:440 (sofia/external/+14792460547 >> @flowroute.com ) State >> Change CS_REPORTING -> CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:749 >> (sofia/external/+14792460547 at flowroute.com >> ) State REPORTING going to sleep >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 >> sofia/external/+14792460547 at flowroute.com >> Standard REPORTING, cause: >> NORMAL_CLEARING >> 2013-01-10 13:39:53.066135 [DEBUG] >> switch_core_state_machine.c:749 >> (sofia/external/+14792460547 at flowroute.com >> ) State REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State Change >> CS_REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 >> Send signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:53.066135 [DEBUG] >> switch_core_state_machine.c:446 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change CS_HANGUP -> >> CS_REPORTING >> >> 2013-01-10 13:39:53.066135 [DEBUG] >> switch_core_state_machine.c:599 Hangup Command with no Session >> system(/usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php >> email=robert.l.craig at craignetwork.com >> extension=3150 >> name=+14792460547-2013-01-10-13-39-26 messages='result: >> Disconnected after permitted retries sender:ATW-4792460547 >> pages:0' domain=192.168.4.15 caller_id_name='' >> caller_id_number=): >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_state_machine.c:667 >> (sofia/external/+14792460547 at flowroute.com >> ) State HANGUP going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 >> sofia/external/+14792460547 at flowroute.com >> Standard HANGUP, cause: >> NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to >> sofia/external/+14792460547 at flowroute.com >> >> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel >> sofia/external/+14792460547 at flowroute.com >> hanging up, cause: >> NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_state_machine.c:667 >> (sofia/external/+14792460547 at flowroute.com >> ) State HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State Change CS_HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com >> ) State EXECUTE going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 >> sofia/external/+14792460547 at flowroute.com >> skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 >> Send signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send >> signal sofia/external/+14792460547 at flowroute.com >> [KILL] >> 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup >> sofia/external/+14792460547 at flowroute.com >> [CS_EXECUTE] [NORMAL_CLEARING] >> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 >> (sofia/external/+14792460547 at flowroute.com >> ) Callstate Change ACTIVE -> >> HANGUP >> EXECUTE sofia/external/+14792460547 at flowroute.com >> hangup() >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 >> sofia/external/+14792460547 at flowroute.com >> Restore previous codec PCMU:0. >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 >> ============================================================================== >> >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote >> model: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote >> vendor: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote >> country: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM >> status off >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer >> Rate: 14400 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image >> resolution: 0x0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total >> fax pages: 0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages >> transferred: 0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local >> station id: SpanDSP Fax Ident >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote >> station id: ATW-4792460547 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax >> processing not successful - result (48) Disconnected after >> permitted retries. >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 >> ============================================================================== >> >> 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct >> ip/port confirmed. >> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 >> sofia/external/+14792460547 at flowroute.com >> image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote >> address:port [66.42.121.5:31632 ] has >> not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote >> address:port [66.42.121.5:31632 ] has >> not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 >> sofia/external/+14792460547 at flowroute.com >> image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote >> address:port [66.42.121.5:31632 ] has >> not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote >> address:port [66.42.121.5:31632 ] has >> not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 >> sofia/external/+14792460547 at flowroute.com >> Set rtp dtmf delay to 40 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 >> dtmf receive payload to 101 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 >> dtmf send payload to 101 >> 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid >> Jitterbuffer spec [0] must be between 20 and 10000 >> 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting >> timer [soft] 160 bytes per 20ms >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP >> [sofia/external/+14792460547 at flowroute.com >> ] 192.168.4.15 port 27588 -> >> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel >> sofia/external/+14792460547 at flowroute.com >> entering state [ready][200] >> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> >> a=T38FaxRateManagement:transferredTCF >> m=image 31632 udptl t38 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849876 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel >> sofia/external/+14792460547 at flowroute.com >> entering state [completing][200] >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 >> sofia/external/+14792460547 at flowroute.com >> Update Callee ID to >> "+14792460547 " <+14792460547 >> > >> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel >> sofia/external/+14792460547 at flowroute.com >> entering state [calling][0] >> 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 >> sofia/external/+14792460547 at flowroute.com >> image media sdp: >> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw >> write codec activation Success L16 >> 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 >> sofia/external/+14792460547 at flowroute.com >> Push codec L16:70 >> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw >> read codec activation Success L16 20000 >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) >> >> 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 >> done playing file silence_stream://2000 >> 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct >> ip/port confirmed. >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel >> sofia/external/+14792460547 at flowroute.com >> entering state [ready][200] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 >> Codec Activated L16 at 8000hz 1 channels 20ms >> EXECUTE sofia/external/+14792460547 at flowroute.com >> playback(silence_stream://2000) >> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET >> [last_fax]=[+14792460547-2013-01-10-13-39-26] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> set(last_fax=+14792460547-2013-01-10-13-39-26) >> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET >> [fax_enable_t38_request]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> set(fax_enable_t38_request=true) >> 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET [fax_enable_t38]=[true] >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel >> sofia/external/+14792460547 at flowroute.com >> entering state [completed][200] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> set(fax_enable_t38=true) >> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel >> [sofia/external/+14792460547 at flowroute.com >> ] has been answered >> 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 >> (sofia/external/+14792460547 at flowroute.com >> ) Callstate Change RINGING -> >> ACTIVE >> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> >> a=sendrecv >> a=ptime:20 >> a=silenceSupp:off - - - - >> a=fmtp:101 0-16 >> a=rtpmap:101 telephone-event/8000 >> a=rtpmap:0 PCMU/8000 >> m=audio 27588 RTP/AVP 0 101 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP >> sofia/external/+14792460547 at flowroute.com >> : >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 >> sofia/external/+14792460547 at flowroute.com >> Set rtp dtmf delay to 40 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 >> dtmf receive payload to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 >> dtmf send payload to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting >> timer [soft] 160 bytes per 20ms >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP >> [sofia/external/+14792460547 at flowroute.com >> ] 192.168.4.15 port 27588 -> >> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success >> [72.201.112.131]:[27588] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> answer() >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET [api_hangup_hook]=[system >> /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php >> email=robert.l.craig at craignetwork.com >> extension=3150 >> name=${last_fax} messages='result: ${fax_result_text} >> sender:${fax_remote_station_id} >> pages:${fax_document_total_pages}' domain=192.168.4.15 >> caller_id_name='${caller_id_name}' >> caller_id_number=${caller_id_number}] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> set(api_hangup_hook=system >> /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php >> email=robert.l.craig at craignetwork.com >> extension=3150 >> name=\${last_fax} messages='result: \${fax_result_text} >> sender:\${fax_remote_station_id} >> pages:\${fax_document_total_pages}' domain=192.168.4.15 >> caller_id_name='\${caller_id_name}' >> caller_id_number=\${caller_id_number}) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET [RFC2822_DATE]=[Thu, 10 >> Jan 2013 13:39:25 -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> set(RFC2822_DATE=Thu, 10 Jan >> 2013 13:39:25 -0700) >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) >> >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> hash(insert/192.168.4.15-spymap/+14792460547 >> /d276f012-5b65-11e2-beae-8564ff827a25) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET [call_direction]=[local] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> set(call_direction=local) >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:209 >> sofia/external/+14792460547 at flowroute.com >> Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 >> sofia/external/+14792460547 at flowroute.com >> SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com >> ) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State Change CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com >> ) State ROUTING going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 >> Send signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:167 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change CS_ROUTING -> >> CS_EXECUTE >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action hangup() >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action >> playback(silence_stream://2000) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action >> set(fax_enable_t38_request=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action set(fax_enable_t38=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action answer() >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action >> set(api_hangup_hook=system /usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php >> email=robert.l.craig at craignetwork.com >> extension=3150 >> name=\\\${last_fax} messages='result: \\\${fax_result_text} >> sender:\\\${fax_remote_station_id} >> pages:\\\${fax_document_total_pages}' domain=192.168.4.15 >> caller_id_name='\\\${caller_id_name}' >> caller_id_number=\\\${caller_id_number}) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (PASS) [Test Fax] >> destination_number(3150) =~ /^3150$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing [default->Test Fax] >> continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [vmain_user] >> destination_number(3150) =~ /^\*97$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing [default->vmain_user] >> continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [vmain] >> destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing [default->vmain] >> continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [send_to_voicemail] destination_number(3150) =~ >> /^\*99(\d{2,7})$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing >> [default->send_to_voicemail] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [extension-intercom] destination_number(3150) =~ >> /^\*8(\d{2,7})$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing >> [default->extension-intercom] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [intercept-ext] >> destination_number(3150) =~ /^\*\*(\d+)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing >> [default->intercept-ext] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [call_return] >> destination_number(3150) =~ /^\*69$|^lcr$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing [default->call_return] >> continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [call_privacy] >> destination_number(3150) =~ /^\*67(\d+)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing >> [default->call_privacy] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action >> hash(insert/${domain_name}-last_dial/global/${uuid}) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) >> >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Absolute Condition [global] >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [global] >> ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing [default->global] >> continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [redial] >> destination_number(3150) =~ /^(redial|\*870)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing [default->redial] >> continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [group-intercept] >> destination_number(3150) =~ /^\*8$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing >> [default->group-intercept] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> ANTI-Action >> set(call_direction=local) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [call-direction] >> ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing >> [default->call-direction] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [Flowroute.11d] >> destination_number(3150) =~ /^\+?(\d{11})$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing >> [default->Flowroute.11d] continue=false >> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 >> Processing +14792460547 <+14792460547 >> >->3150 in context default >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:117 >> sofia/external/+14792460547 at flowroute.com >> Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 >> sofia/external/+14792460547 at flowroute.com >> SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com >> ) State ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State Change CS_ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com >> ) State EXECUTE going to sleep >> 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer >> sofia/external/+14792460547 at flowroute.com >> to XML[3150 at default] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 >> Send signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change CS_EXECUTE -> >> CS_ROUTING >> EXECUTE sofia/external/+14792460547 at flowroute.com >> transfer(3150 XML default) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT >> (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> export(RFC2822_DATE=Thu, 10 >> Jan 2013 13:39:25 -0700) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET [outside_call]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> set(outside_call=true) >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:209 >> sofia/external/+14792460547 at flowroute.com >> Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 >> sofia/external/+14792460547 at flowroute.com >> SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com >> ) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State Change CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com >> ) State ROUTING going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 >> Send signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:167 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change CS_ROUTING -> >> CS_EXECUTE >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action transfer(3150 XML default) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (PASS) [Test_Number] >> destination_number(15203039810 ) =~ >> /^15203039810$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (PASS) [Test_Number] >> context(public) =~ /public/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing [public->Test_Number] >> continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [call_debug] >> ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing [public->call_debug] >> continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Action set(outside_call=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Absolute Condition [outside_call] >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing [public->outside_call] >> continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (FAIL) [unloop] >> ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> Regex (PASS) [unloop] >> ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com >> parsing [public->unloop] >> continue=false >> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 >> Processing +14792460547 <+14792460547 >> >->15203039810 in context >> public >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:117 >> sofia/external/+14792460547 at flowroute.com >> Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 >> sofia/external/+14792460547 at flowroute.com >> SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com >> ) State ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 >> (sofia/external/+14792460547 at flowroute.com >> ) Callstate Change DOWN -> RINGING >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State Change CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_core_state_machine.c:454 >> (sofia/external/+14792460547 at flowroute.com >> ) State INIT going to sleep >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 >> Send signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change CS_INIT -> >> CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 >> sofia/external/+14792460547 at flowroute.com >> SOFIA INIT >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_core_state_machine.c:454 >> (sofia/external/+14792460547 at flowroute.com >> ) State INIT >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State Change CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 >> Send signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change CS_NEW -> CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 >> dtmf send/recv payload to 101 >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 >> sofia/external/+14792460547 at flowroute.com >> Original read codec set to PCMU:0 >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec >> sofia/external/+14792460547 at flowroute.com >> PCMU/8000 20 ms 160 samples >> 64000 bits >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec >> Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec >> Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >> >> a=ptime:20 >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:18 G729/8000 >> m=audio 31632 RTP/AVP 0 18 101 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849875 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel >> sofia/external/+14792460547 at flowroute.com >> entering state [received][100] >> 2013-01-10 13:39:25.786370 [DEBUG] >> switch_core_state_machine.c:433 >> (sofia/external/+14792460547 at flowroute.com >> ) State NEW >> 2013-01-10 13:39:25.786370 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State Change CS_NEW >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New >> Channel sofia/external/+14792460547 at flowroute.com >> >> [d276f012-5b65-11e2-beae-8564ff827a25] >> >> Robert >> >> >> >> ------------------------------------------------------------------------ >> Date: Thu, 10 Jan 2013 09:47:18 -0800 >> From: msc at freeswitch.org >> To: freeswitch-users at lists.freeswitch.org >> >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Can you pastebin a console debug log w/ sip trace of a call >> failing to negotiate t.38 correctly? That might help us figure >> out where to look. >> -MC >> >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig >> > wrote: >> >> Just installed FusionPBX using the script built for a Centos >> 6 install. Everything works great in regards to calls, but my >> main goal of this project was to utilize the fax capabilities >> of Freeswitch/FusionPBX. I have tried many things in media >> bypass, routing to a fax extension, using an ATA, etc., but I >> can't get T38 to negotiate correctly. Can someone point me in >> the right direction as to where I should begin? I am guessing >> it isn't supposed to work right out of the box. I am using >> Flowroute as my provider right now and the have confirmed >> that they utilize T38 in both directions. Thanks! >> >> >> Robert >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official >> FreeSWITCH Sites http://www.freeswitch.org >> http://wiki.freeswitch.org http://www.cluecon.com >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official >> FreeSWITCH Sites http://www.freeswitch.org >> http://wiki.freeswitch.org http://www.cluecon.com >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/ccef5aac/attachment-0001.html From jackal at cybershroud.net Mon Jan 14 18:13:34 2013 From: jackal at cybershroud.net (Carlos Flor) Date: Mon, 14 Jan 2013 10:13:34 -0500 Subject: [Freeswitch-users] Chatplan: is msg received success? In-Reply-To: References: Message-ID: Alex, You need to set blocking=true in the chatplan. This will include a Delivery-Failure=true/false header in the event that is generated when you send the message. You will need to have an ESL client parse that field to determine next steps. If you don't explicitly set blocking to true, then FS doesn't wait for a response and so it has no way of knowing if the message was delivered or not. For that reason, the Delivery-Failure header is not present in the event when in non-blocking mode. On Mon, Jan 14, 2013 at 6:53 AM, Pip Live wrote: > Hi, I use Freeswitch to handle SIP message between SIP extentions, how can > server know does the message be received by client or not? I noticed even > if a SIP extention is offline/unregister, SIP message will still be sent by > server but just the receiver will never get it (even if later the receiver > online/registered again). > > What I want to achieve is: send SIP message to an extention, if received > OK, nothing happen; if not, forward the message to user's email address. > > brgds, Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/5ccea19d/attachment.html From steveayre at gmail.com Mon Jan 14 18:14:18 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 14 Jan 2013 15:14:18 +0000 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: References: Message-ID: What exactly would you be 'testing' though? There's no licensed version of the module available yet. >From the Wiki: Licensing is being worked on and FreeSWITCH? may be able to offer licensing terms in the future for this module. If you're interested in a transcoding version of the AMR codec for production use I'd suggest you contact the core developers to either put your name down for how many licenses you'd be interested in, or raise a bounty (like happened with G729). -Steve On 14 January 2013 13:04, Luis Azedo wrote: > Hi, > is there any way to enable mod_amr with transcoding for testing purposes ? > maybe with opencore-amr ? > thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/b40ec554/attachment.html From sdevoy at bizfocused.com Mon Jan 14 18:29:02 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 14 Jan 2013 10:29:02 -0500 Subject: [Freeswitch-users] Delay Round Two...FIGHT! In-Reply-To: References: Message-ID: <0ed201cdf26b$e1e9c9b0$a5bd5d10$@bizfocused.com> I had this EXACT problem on a single (or maybe dual??) core, single CPU FS box. I upgraded to QUAD core system and the problem went away immediately. Mine was not on every call, but if there were just a few concurrent calls it would happen. That is what led me to the "lack of resources" conclusion. In my case you could see in the FS log a 5 second or longer delay in the actual connection of the RTP. I have seen people give stats about FS handling thousands of calls per second. I have often wondered if they checked to see how long it took to actually connect the RTP channels in those tests. I am highly skeptical. But I have no issues on a single QUAD core system. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Blake Priddy Sent: Monday, January 14, 2013 9:46 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Delay Round Two...FIGHT! Ok everyone, hope your new years went wonderful! So here is the situation again. THE DELAY IS BACK! I have wiped FS and installed the latest git head and the delay is still present. I want to eliminate all FS options that could fix it before I call AT&T. Any help is appreciated! Person from outside the school calls, secretary answers, secretary hears they person saying hello, secretary says hello over and over again for 10 seconds. Then the B-Leg audio finally is heard by the A-LEG. Note* The B-Leg (secretary) can hear the A-leg talking the whole time. Just the A-Leg cannot hear the (secretary) B-Leg until about 7-13 seconds has passed. I see nothing in the FS log after the audio from the B-Leg finally traverses to the A-Leg. Please help our school fellow FS users! LONG LIVE FS!!! :) -- Blakelund Priddy Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/36931008/attachment.html From william.king at quentustech.com Mon Jan 14 18:47:29 2013 From: william.king at quentustech.com (William King) Date: Mon, 14 Jan 2013 07:47:29 -0800 Subject: [Freeswitch-users] Delay Round Two...FIGHT! In-Reply-To: References: Message-ID: <50F42891.7090000@quentustech.com> Do you happen to be running the stable branch? If so then I believe the issue you are hitting was already fixed in master. If you have a chance to test, see if the length of the audio delay is about the same amount of time as the secretaries phone was ringing. If that is the case, then it should already be fixed in master. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 01/14/2013 06:45 AM, Blake Priddy wrote: > Ok everyone, hope your new years went wonderful! So here is the > situation again. THE DELAY IS BACK! I have wiped FS and installed the > latest git head and the delay is still present. I want to eliminate all > FS options that could fix it before I call AT&T. Any help is > appreciated! Person from outside the school calls, secretary answers, > secretary hears they person saying hello, secretary says hello over and > over again for 10 seconds. Then the B-Leg audio finally is heard by the > A-LEG. Note* The B-Leg (secretary) can hear the A-leg talking the whole > time. Just the A-Leg cannot hear the (secretary) B-Leg until about 7-13 > seconds has passed. I see nothing in the FS log after the audio from the > B-Leg finally traverses to the A-Leg. Please help our school fellow FS > users! > > LONG LIVE FS!!! :) > > -- > > *Blakelund Priddy* > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From roger.castaldo at gmail.com Mon Jan 14 19:10:22 2013 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Mon, 14 Jan 2013 11:10:22 -0500 Subject: [Freeswitch-users] Delay Round Two...FIGHT! In-Reply-To: <50F42891.7090000@quentustech.com> References: <50F42891.7090000@quentustech.com> Message-ID: Another option, to at least eliminate issues, assuming it happens on all calls is to connect a phone either directly to the server, ie spare nic port or as close as possible, ie same switch and perform an external call to that phone to see if the delay occurs. This would at least allow you to eliminate a network latency between the server and the secretaries phone which would cause delays in audio as well since freeswitch cannot send audio if the audio hasn't arrived to it yet. Just a thought on trying to eliminate the possible. On Mon, Jan 14, 2013 at 10:47 AM, William King wrote: > Do you happen to be running the stable branch? If so then I believe the > issue you are hitting was already fixed in master. If you have a chance > to test, see if the length of the audio delay is about the same amount > of time as the secretaries phone was ringing. If that is the case, then > it should already be fixed in master. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 01/14/2013 06:45 AM, Blake Priddy wrote: > > Ok everyone, hope your new years went wonderful! So here is the > > situation again. THE DELAY IS BACK! I have wiped FS and installed the > > latest git head and the delay is still present. I want to eliminate all > > FS options that could fix it before I call AT&T. Any help is > > appreciated! Person from outside the school calls, secretary answers, > > secretary hears they person saying hello, secretary says hello over and > > over again for 10 seconds. Then the B-Leg audio finally is heard by the > > A-LEG. Note* The B-Leg (secretary) can hear the A-leg talking the whole > > time. Just the A-Leg cannot hear the (secretary) B-Leg until about 7-13 > > seconds has passed. I see nothing in the FS log after the audio from the > > B-Leg finally traverses to the A-Leg. Please help our school fellow FS > > users! > > > > LONG LIVE FS!!! :) > > > > -- > > > > *Blakelund Priddy* > > Network Systems Engineer > > Bryant Public School District > > Bryant, Arkansas 72022 > > http://www.bryantschools.org > > p 501-653-5038 > > f 501-847-5656 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/ec8a840e/attachment-0001.html From luis.azedo at factorlusitano.com Mon Jan 14 19:27:14 2013 From: luis.azedo at factorlusitano.com (Luis Azedo) Date: Mon, 14 Jan 2013 16:27:14 +0000 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes Message-ID: i know there is no licensed version of the module available yet. i have a dialer that supports amr and want to see if it can be used in freeswitch with transcoding. like in here http://sourceforge.net/projects/asterisk-amr/ but for freeswitch does anyone have mod_amr with transcoding running in freeswitch ? -- Luis Azedo From bpriddy at bryantschools.org Mon Jan 14 19:32:47 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Mon, 14 Jan 2013 10:32:47 -0600 Subject: [Freeswitch-users] Delay Round Two...FIGHT! In-Reply-To: References: <50F42891.7090000@quentustech.com> Message-ID: Well our server has 6 cores. William, we are running stable on debian. Any other brain busters? I need as many possibilites of issues as possible. Roger, we have gig across the network. But I will give it a shot just to eliminate that possibility. On Mon, Jan 14, 2013 at 10:10 AM, Roger Castaldo wrote: > Another option, to at least eliminate issues, assuming it happens on all > calls is to connect a phone either directly to the server, ie spare nic > port or as close as possible, ie same switch and perform an external call > to that phone to see if the delay occurs. This would at least allow you to > eliminate a network latency between the server and the secretaries phone > which would cause delays in audio as well since freeswitch cannot send > audio if the audio hasn't arrived to it yet. Just a thought on trying to > eliminate the possible. > > > On Mon, Jan 14, 2013 at 10:47 AM, William King < > william.king at quentustech.com> wrote: > >> Do you happen to be running the stable branch? If so then I believe the >> issue you are hitting was already fixed in master. If you have a chance >> to test, see if the length of the audio delay is about the same amount >> of time as the secretaries phone was ringing. If that is the case, then >> it should already be fixed in master. >> >> William King >> Senior Engineer >> Quentus Technologies, INC >> 1037 NE 65th St Suite 273 >> Seattle, WA 98115 >> Main: (877) 211-9337 >> Office: (206) 388-4772 >> Cell: (253) 686-5518 >> william.king at quentustech.com >> >> On 01/14/2013 06:45 AM, Blake Priddy wrote: >> > Ok everyone, hope your new years went wonderful! So here is the >> > situation again. THE DELAY IS BACK! I have wiped FS and installed the >> > latest git head and the delay is still present. I want to eliminate all >> > FS options that could fix it before I call AT&T. Any help is >> > appreciated! Person from outside the school calls, secretary answers, >> > secretary hears they person saying hello, secretary says hello over and >> > over again for 10 seconds. Then the B-Leg audio finally is heard by the >> > A-LEG. Note* The B-Leg (secretary) can hear the A-leg talking the whole >> > time. Just the A-Leg cannot hear the (secretary) B-Leg until about 7-13 >> > seconds has passed. I see nothing in the FS log after the audio from the >> > B-Leg finally traverses to the A-Leg. Please help our school fellow FS >> > users! >> > >> > LONG LIVE FS!!! :) >> > >> > -- >> > >> > *Blakelund Priddy* >> > Network Systems Engineer >> > Bryant Public School District >> > Bryant, Arkansas 72022 >> > http://www.bryantschools.org >> > p 501-653-5038 >> > f 501-847-5656 >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/9ebc03a7/attachment.html From william.king at quentustech.com Mon Jan 14 19:39:26 2013 From: william.king at quentustech.com (William King) Date: Mon, 14 Jan 2013 08:39:26 -0800 Subject: [Freeswitch-users] Delay Round Two...FIGHT! In-Reply-To: References: <50F42891.7090000@quentustech.com> Message-ID: <50F434BE.9060602@quentustech.com> If you have a chance try the master branch. I'd almost bet the issue will be gone. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 01/14/2013 08:32 AM, Blake Priddy wrote: > Well our server has 6 cores. William, we are running stable on debian. > Any other brain busters? I need as many possibilites of issues as > possible. Roger, we have gig across the network. But I will give it a > shot just to eliminate that possibility. > > > On Mon, Jan 14, 2013 at 10:10 AM, Roger Castaldo > > wrote: > > Another option, to at least eliminate issues, assuming it happens on > all calls is to connect a phone either directly to the server, ie > spare nic port or as close as possible, ie same switch and perform > an external call to that phone to see if the delay occurs. This > would at least allow you to eliminate a network latency between the > server and the secretaries phone which would cause delays in audio > as well since freeswitch cannot send audio if the audio hasn't > arrived to it yet. Just a thought on trying to eliminate the possible. > > > On Mon, Jan 14, 2013 at 10:47 AM, William King > > > wrote: > > Do you happen to be running the stable branch? If so then I > believe the > issue you are hitting was already fixed in master. If you have a > chance > to test, see if the length of the audio delay is about the same > amount > of time as the secretaries phone was ringing. If that is the > case, then > it should already be fixed in master. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 01/14/2013 06:45 AM, Blake Priddy wrote: > > Ok everyone, hope your new years went wonderful! So here is the > > situation again. THE DELAY IS BACK! I have wiped FS and > installed the > > latest git head and the delay is still present. I want to > eliminate all > > FS options that could fix it before I call AT&T. Any help is > > appreciated! Person from outside the school calls, secretary > answers, > > secretary hears they person saying hello, secretary says hello > over and > > over again for 10 seconds. Then the B-Leg audio finally is > heard by the > > A-LEG. Note* The B-Leg (secretary) can hear the A-leg talking > the whole > > time. Just the A-Leg cannot hear the (secretary) B-Leg until > about 7-13 > > seconds has passed. I see nothing in the FS log after the > audio from the > > B-Leg finally traverses to the A-Leg. Please help our school > fellow FS > > users! > > > > LONG LIVE FS!!! :) > > > > -- > > > > *Blakelund Priddy* > > Network Systems Engineer > > Bryant Public School District > > Bryant, Arkansas 72022 > > http://www.bryantschools.org > > p 501-653-5038 > > f 501-847-5656 > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > > *Blakelund Priddy* > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Mon Jan 14 19:45:04 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 14 Jan 2013 11:45:04 -0500 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: References: Message-ID: <439D90B8-16C2-457C-82EB-524D179A2486@jerris.com> Licensing for that pool last we looked was a bit of a mess IIRC. We will need a lot of interest to consider pursuing this. That all being said, the module in tree is complete module code, just requires the library. Makefile: AMR_DIR=$(switch_srcdir)/libs/amr AMR_BUILDDIR=$(switch_builddir)/libs/amr AMR_A=$(AMR_BUILDDIR)/libamr.a LOCAL_INSERT_CFLAGS=if test -d $(AMR_DIR); then echo "-I$(AMR_DIR)" ; else echo "-DAMR_PASSTHROUGH" ; fi ; LOCAL_INSERT_LDFLAGS=test ! -d $(AMR_DIR) || echo "$(AMR_A)"; if you put the correct lib in libs/amr and build it then build the module, it should work. your looking for a lib with these header files: #include "interf_enc.h" #include "interf_dec.h" we do not distribute or provide any licensing for this library, so you will need to arrange that on your own. Mike On Jan 14, 2013, at 10:14 AM, Steven Ayre wrote: > What exactly would you be 'testing' though? There's no licensed version of the module available yet. > > From the Wiki: > Licensing is being worked on and FreeSWITCH? may be able to offer licensing terms in the future for this module. > > If you're interested in a transcoding version of the AMR codec for production use I'd suggest you contact the core developers to either put your name down for how many licenses you'd be interested in, or raise a bounty (like happened with G729). > > -Steve > > > > > On 14 January 2013 13:04, Luis Azedo wrote: > Hi, > is there any way to enable mod_amr with transcoding for testing purposes ? > maybe with opencore-amr ? > thanks > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/b46ab964/attachment-0001.html From raimund.sacherer at logitravel.com Mon Jan 14 20:36:09 2013 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Mon, 14 Jan 2013 18:36:09 +0100 (CET) Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: <23602598.3790.1358184950672.JavaMail.javamailuser@localhost> Message-ID: <15959415.3795.1358184967040.JavaMail.javamailuser@localhost> Thank you Dmitry, are you saying you only use the opensource capture part from voip-monitor on all your 10 FS servers and you do not use their webinterface at all? If I understood this correctly, this opensource does all the capturing and it calculates MOS and Loss scores and stores it in a central MySQL database? That sound's great, I thought the MOS and Loss calculations are in their proprietery part, but if it's included in the opensource part it could be interesting to create your own web-visualization. On the other hand their licence seems to be fairly cheap, some 480 euros ... but I am not sure if this is per server you monitor, or per server you deploy the web-interface to. best, Rai ----- Original Message ----- From: "Dmitry Sytchev" To: "FreeSWITCH Users Help" Sent: Viernes, 11 de Enero 2013 14:35:33 Subject: Re: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension We use VoipMonitor open-source part, with homebrew custom WEB-UI. This is a really cool software, it continuously evolves and captures pretty well even large amounts of VoiP traffic. I highly recommend this software, since it doesn't only captures signaling traffic, but also can capture media/rtcp, automatically convert captured calls to wav files or store raw pcaps for further processing. it also can analyze MOS/Loss scores and many many other different parameters and store them in database per call. We use single instance to store data for investigation of calls with problems on more than 10 servers. 2013/1/11 Raimund Sacherer < raimund.sacherer at logitravel.com > Does anyone have any experience with voip monitor? http://www.voipmonitor.org/ We will start evaluating it in our environment as we are switching more and more to SIP providers, therefore I would love to have better Monitoring. What voipmonitor says is that they are able to filter the calls in realtime with low overhead an if call quality drops, they can alert you, and they seem to have a rather nice web interface and they also are used by others to store the RTP's ... What seems to be especially nice is that their capture daemon (as they state it) adds very little overhead to a system, so depending on call quantity they say it's neglect-able. Would love to hear some comparisions between this and HOMER if some can provide it. Thank you, best Ray From: "Cal Leeming [Simplicity Media Ltd]" < cal.leeming at simplicitymedialtd.co.uk > To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > Sent: Domingo, 6 de Enero 2013 21:01:24 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension Could a core dev speculate on how much the expected bounty would be for this? Thanks Cal _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 S?guenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para m?vil Logitravel.com Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electr?nico por error, se ruega notif?quese inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formar?n parte de un fichero registrado como CLIENTES con n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a estos datos que deber? ejercer mediante escrito a la direcci?n anteriormente citada. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Best regards, Dmitry Sytchev, IT Engineer _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/ff55a34e/attachment.html From steveayre at gmail.com Mon Jan 14 20:40:12 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 14 Jan 2013 17:40:12 +0000 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: <439D90B8-16C2-457C-82EB-524D179A2486@jerris.com> References: <439D90B8-16C2-457C-82EB-524D179A2486@jerris.com> Message-ID: An alternative is mod_sangoma_codec, which supports AMR. This is the module to interface with a transcoding hardware card made by Sangoma, which implements several codecs including licensed ones such as AMR, G729 etc. The license cost is included in the cost of the hardware itself. The transcoding is done on the card itself, offloading that processing from the CPU. Since mod_amr licenses don't exist yet, that's probably your only legal option for AMR transcoding in FreeSWITCH right now. With a bit of research you can (quite easily) find the AMR library mod_amr is designed for from the headers Mike mentioned, but you won't get any help from the community finding, implementing or debugging it due to the legal issues surrounding doing so. As for testing with mod_amr... testing one codec implementation tells you exactly nothing about a different implementation. If you plan to test transcoding AMR you *need* to test with the implementation you will use in production to get any meaningful results. Which won't be mod_amr. Suffice to say transcoding in FreeSWITCH works very well, providing the in/out codecs are supported. -Steve On 14 January 2013 16:45, Michael Jerris wrote: > Licensing for that pool last we looked was a bit of a mess IIRC. We will > need a lot of interest to consider pursuing this. That all being said, the > module in tree is complete module code, just requires the library. > > Makefile: > AMR_DIR=$(switch_srcdir)/libs/amr > AMR_BUILDDIR=$(switch_builddir)/libs/amr > AMR_A=$(AMR_BUILDDIR)/libamr.a > > LOCAL_INSERT_CFLAGS=if test -d $(AMR_DIR); then echo "-I$(AMR_DIR)" ; else > echo "-DAMR_PASSTHROUGH" ; fi ; > LOCAL_INSERT_LDFLAGS=test ! -d $(AMR_DIR) || echo "$(AMR_A)"; > > > if you put the correct lib in libs/amr and build it then build the module, > it should work. > > your looking for a lib with these header files: > > #include "interf_enc.h" > > #include "interf_dec.h" > > > we do not distribute or provide any licensing for this library, so you > will need to arrange that on your own. > > Mike > > > On Jan 14, 2013, at 10:14 AM, Steven Ayre wrote: > > What exactly would you be 'testing' though? There's no licensed version of > the module available yet. > > From the Wiki: > Licensing is being worked on and FreeSWITCH? may be able to offer > licensing terms in the future for this module. > > If you're interested in a transcoding version of the AMR codec for > production use I'd suggest you contact the core developers to either put > your name down for how many licenses you'd be interested in, or raise a > bounty (like happened with G729). > > -Steve > > > > > On 14 January 2013 13:04, Luis Azedo wrote: > >> Hi, >> is there any way to enable mod_amr with transcoding for testing purposes >> ? >> maybe with opencore-amr ? >> thanks >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/3e45e7c4/attachment-0001.html From schoch+freeswitch.org at xwin32.com Mon Jan 14 21:17:05 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 14 Jan 2013 10:17:05 -0800 Subject: [Freeswitch-users] sources for SIP hotline phone stations In-Reply-To: <002d01cdf074$5a66ede0$0f34c9a0$@com> References: <022b01cdf011$48d71d60$da855820$@com> <50F0564D.4020507@puzzled.xs4all.nl> <002d01cdf074$5a66ede0$0f34c9a0$@com> Message-ID: On Fri, Jan 11, 2013 at 7:24 PM, Bote Man wrote: > I suppose that then I'd need to find a supplier of red handsets, or get > a can of red paint. :- ) Don't discount that idea. Lately I've been watching The Ben Heck Show ( http://www.element14.com/community/community/experts/benheck). Many of the shows are all about taking some piece of electronics (often a video game), and repackaging it, usually smaller and portable. If you already have a supply of good looking red analog phones, it wouldn't be that difficult to rip the insides out of a VoIP phone, ditch the keypad, and repackage it inside the old red phone. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/e83fa2e3/attachment.html From kbdfck at gmail.com Mon Jan 14 21:34:06 2013 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Mon, 14 Jan 2013 22:34:06 +0400 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: <15959415.3795.1358184967040.JavaMail.javamailuser@localhost> References: <23602598.3790.1358184950672.JavaMail.javamailuser@localhost> <15959415.3795.1358184967040.JavaMail.javamailuser@localhost> Message-ID: We started development of web-ui rather as programming exercise in JS, twitter bootstrap and perl Mojolicious :) Yes, engine itself does all stuff we need to analyze calls, it stores data in db, can be set up to use multiple sensors and store their trace data in same DB. Obviously, pcaps and other large raw media data is stored locally on sensor machines. We use it to do full monitoring of all legs of all VoIP calls in our network. 2013/1/14 Raimund Sacherer > Thank you Dmitry, > > are you saying you only use the opensource capture part from voip-monitor > on all your 10 FS servers and you do not use their webinterface at all? If > I understood this correctly, this opensource does all the capturing and it > calculates MOS and Loss scores and stores it in a central MySQL database? > > That sound's great, I thought the MOS and Loss calculations are in their > proprietery part, but if it's included in the opensource part it could be > interesting to create your own web-visualization. On the other hand their > licence seems to be fairly cheap, some 480 euros ... but I am not sure if > this is per server you monitor, or per server you deploy the web-interface > to. > > best, > Rai > > ------------------------------ > *From: *"Dmitry Sytchev" > > *To: *"FreeSWITCH Users Help" > *Sent: *Viernes, 11 de Enero 2013 14:35:33 > *Subject: *Re: [Freeswitch-users] Bounty for - FS-4268 Sip trace with > filter for IP or extension > > > We use VoipMonitor open-source part, with homebrew custom WEB-UI. This is > a really cool software, it continuously evolves and captures pretty well > even large amounts of VoiP traffic. > > I highly recommend this software, since it doesn't only captures signaling > traffic, but also can capture media/rtcp, automatically convert captured > calls to wav files or store raw pcaps for further processing. it also can > analyze MOS/Loss scores and many many other different parameters and store > them in database per call. > > We use single instance to store data for investigation of calls with > problems on more than 10 servers. > > > > > 2013/1/11 Raimund Sacherer > >> Does anyone have any experience with voip monitor? >> http://www.voipmonitor.org/ >> >> We will start evaluating it in our environment as we are switching more >> and more to SIP providers, therefore I would love to have better >> Monitoring. >> >> What voipmonitor says is that they are able to filter the calls in >> realtime with low overhead an if call quality drops, they can alert you, >> and they seem to have a rather nice web interface and they also are used by >> others to store the RTP's ... >> >> What seems to be especially nice is that their capture daemon (as they >> state it) adds very little overhead to a system, so depending on call >> quantity they say it's neglect-able. >> >> Would love to hear some comparisions between this and HOMER if some can >> provide it. >> >> Thank you, best >> Ray >> >> ------------------------------ >> *From: *"Cal Leeming [Simplicity Media Ltd]" < >> cal.leeming at simplicitymedialtd.co.uk> >> *To: *"FreeSWITCH Users Help" >> *Sent: *Domingo, 6 de Enero 2013 21:01:24 >> *Subject: *[Freeswitch-users] Bounty for - FS-4268 Sip trace with filter >> for IP or extension >> >> >> Could a core dev speculate on how much the expected bounty would be for >> this? >> >> Thanks >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> [image: Logitravel.com] Raimund Sacherer >> *Sistemas*Agencia de Viajes Online >> www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) >> Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca >> Tel 902 366 847 | Fax 971 213 495 S?guenos en:[image: Facebook de >> Logitravel] [image: Twitter de >> Logitravel] [image: Blog de Logitravel] [image: >> Logitravel en Youtube] [image: >> Logitravel en Foursquare] Descarga >> nuestras *aplicaciones para m?vil* [image: Logitravel.com]Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene >> informaci?n de car?cter confidencial exclusivamente dirigida a su >> destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a >> terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de >> haber recibido este correo electr?nico por error, se ruega notif?quese >> inmediatamente esta circunstancia mediante reenv?o a la direcci?n >> electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus >> datos forman o formar?n parte de un fichero registrado como CLIENTES con >> n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de >> Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio >> Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. >> Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a >> estos datos que deber? ejercer mediante escrito a la direcci?n >> anteriormente citada. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Best regards, > > Dmitry Sytchev, > IT Engineer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Best regards, Dmitry Sytchev, IT Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/c6bc2feb/attachment.html From gabe at gundy.org Mon Jan 14 22:22:55 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 14 Jan 2013 12:22:55 -0700 Subject: [Freeswitch-users] How to pause and resume play an audio file in conference? In-Reply-To: <201301141547320568849@alongtechnology.com.cn> References: <201301141547320568849@alongtechnology.com.cn> Message-ID: On Mon, Jan 14, 2013 at 12:47 AM, ?? wrote: > "uuid_fileman" is for channel, not for a conference. Please trim your emails if you're going to reply to a list digest. It helps keep the noise down and the signal up. Thanks, Gabe From schoch+freeswitch.org at xwin32.com Mon Jan 14 22:27:48 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 14 Jan 2013 11:27:48 -0800 Subject: [Freeswitch-users] Multiple Network Interfaces Message-ID: (Don't hesitate to tell me that I'm doing this the wrong way and what is the right way to do it.) I have 3 network cards on our Freeswitch box (which will be used as the office PBX, as soon as I can figure out how to get this all working.) One card is on our office LAN. It's address is 192.168.1.17. The default route is 192.168.1.9, which is our PepLink Balance 20 router, which knows how to route VoIP, and balances the outgoing network traffic via 2 WANs: an AT&T DSL and one from tlinkbroadband.com. The second card is connected to a PoE switch that will connect to all the desktop phones. It's address is 192.168.4.1. The FS machine runs DHCP, TFTP, FTP, and of course FS on this interface. The third card is not yet configured with an address. It will be an outside address, so that users can connect from home using SIP to check their voicemail, make office calls, etc. It may also be used as a failsafe interface on case the PepLink goes down. The phones connect successfully to FS over the 192.168.4.1 address. FS connects to sip.flowroute.com over the default route. Outgoing calls work. (When I get a DID assigned to our flowroute.com account, and get the auto-attendant working, then I expect it will work correctly in that direction. What doesn't work is SIP FAX software that I just installed on a PC on the LAN. I attempted to connect it using 192.168.1.17, but that didn't work, so I used 192.168.4.1, which is routed by the router to the FS box. The SIP registration was successful, and the test call was actually placed and rang on the receiving FAX machine, but then it failed. I have a suspicion that the audio got dropped somewhere. I also think that if I can get FS to accept a SIP registration on the other network card, that it might work. I can't find anything in the wiki about multiple network cards (except a page that was removed) -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/11b194a3/attachment.html From avi at avimarcus.net Mon Jan 14 22:36:50 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 14 Jan 2013 21:36:50 +0200 Subject: [Freeswitch-users] Multiple Network Interfaces In-Reply-To: References: Message-ID: Just to clarify: do you have one sofia profile per IP:port? -Avi On Mon, Jan 14, 2013 at 9:27 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > (Don't hesitate to tell me that I'm doing this the wrong way and what is > the right way to do it.) > > I have 3 network cards on our Freeswitch box (which will be used as the > office PBX, as soon as I can figure out how to get this all working.) > > One card is on our office LAN. It's address is 192.168.1.17. The default > route is 192.168.1.9, which is our PepLink Balance 20 router, which knows > how to route VoIP, and balances the outgoing network traffic via 2 WANs: an > AT&T DSL and one from tlinkbroadband.com. > > The second card is connected to a PoE switch that will connect to all the > desktop phones. It's address is 192.168.4.1. The FS machine runs DHCP, > TFTP, FTP, and of course FS on this interface. > > The third card is not yet configured with an address. It will be an > outside address, so that users can connect from home using SIP to check > their voicemail, make office calls, etc. It may also be used as a failsafe > interface on case the PepLink goes down. > > The phones connect successfully to FS over the 192.168.4.1 address. FS > connects to sip.flowroute.com over the default route. Outgoing calls > work. (When I get a DID assigned to our flowroute.com account, and get > the auto-attendant working, then I expect it will work correctly in that > direction. > > What doesn't work is SIP FAX software that I just installed on a PC on the > LAN. I attempted to connect it using 192.168.1.17, but that didn't work, > so I used 192.168.4.1, which is routed by the router to the FS box. The > SIP registration was successful, and the test call was actually placed and > rang on the receiving FAX machine, but then it failed. I have a suspicion > that the audio got dropped somewhere. I also think that if I can get FS to > accept a SIP registration on the other network card, that it might work. > > I can't find anything in the wiki about multiple network cards (except a > page that was removed) > > -- > Steve > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/a6bb5795/attachment.html From kris at kriskinc.com Mon Jan 14 22:58:32 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 14 Jan 2013 14:58:32 -0500 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: References: <439D90B8-16C2-457C-82EB-524D179A2486@jerris.com> Message-ID: IIRC Sangoma doesn't provide licensing for AMR and AMR-WB (just the implementation). AFAIK mod_sangoma_codec is no more legal than libamr. On Mon, Jan 14, 2013 at 12:40 PM, Steven Ayre wrote: > An alternative is mod_sangoma_codec, which supports AMR. > > This is the module to interface with a transcoding hardware card made by > Sangoma, which implements several codecs including licensed ones such as > AMR, G729 etc. The license cost is included in the cost of the hardware > itself. The transcoding is done on the card itself, offloading that > processing from the CPU. > > Since mod_amr licenses don't exist yet, that's probably your only legal > option for AMR transcoding in FreeSWITCH right now. > > With a bit of research you can (quite easily) find the AMR library mod_amr > is designed for from the headers Mike mentioned, but you won't get any help > from the community finding, implementing or debugging it due to the legal > issues surrounding doing so. > > As for testing with mod_amr... testing one codec implementation tells you > exactly nothing about a different implementation. If you plan to test > transcoding AMR you *need* to test with the implementation you will use in > production to get any meaningful results. Which won't be mod_amr. > > Suffice to say transcoding in FreeSWITCH works very well, providing the > in/out codecs are supported. > > -Steve > > > > On 14 January 2013 16:45, Michael Jerris wrote: >> >> Licensing for that pool last we looked was a bit of a mess IIRC. We will >> need a lot of interest to consider pursuing this. That all being said, the >> module in tree is complete module code, just requires the library. >> >> Makefile: >> AMR_DIR=$(switch_srcdir)/libs/amr >> AMR_BUILDDIR=$(switch_builddir)/libs/amr >> AMR_A=$(AMR_BUILDDIR)/libamr.a >> >> LOCAL_INSERT_CFLAGS=if test -d $(AMR_DIR); then echo "-I$(AMR_DIR)" ; else >> echo "-DAMR_PASSTHROUGH" ; fi ; >> LOCAL_INSERT_LDFLAGS=test ! -d $(AMR_DIR) || echo "$(AMR_A)"; >> >> >> if you put the correct lib in libs/amr and build it then build the module, >> it should work. >> >> your looking for a lib with these header files: >> >> #include "interf_enc.h" >> #include "interf_dec.h" >> >> we do not distribute or provide any licensing for this library, so you >> will need to arrange that on your own. >> >> Mike >> >> >> On Jan 14, 2013, at 10:14 AM, Steven Ayre wrote: >> >> What exactly would you be 'testing' though? There's no licensed version of >> the module available yet. >> >> From the Wiki: >> Licensing is being worked on and FreeSWITCH? may be able to offer >> licensing terms in the future for this module. >> >> If you're interested in a transcoding version of the AMR codec for >> production use I'd suggest you contact the core developers to either put >> your name down for how many licenses you'd be interested in, or raise a >> bounty (like happened with G729). >> >> -Steve >> >> >> >> >> On 14 January 2013 13:04, Luis Azedo >> wrote: >>> >>> Hi, >>> is there any way to enable mod_amr with transcoding for testing purposes >>> ? >>> maybe with opencore-amr ? >>> thanks >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From jpyle at fidelityvoice.com Mon Jan 14 23:12:06 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Mon, 14 Jan 2013 15:12:06 -0500 Subject: [Freeswitch-users] inbound-proxy-media with attended transfer Message-ID: Hello, Is inbound-proxy-media compatible with attended transfers in any way, shape or form? Is there any way to avoid: [ERR] switch_core_io.c:1151 Codec PROXY PASS-THROUGH encoder error! Is there any way to emulate the behavior of a "proxy-media-after-att-xfer" option? I strongly prefer the codec transparency of inbound-proxy-media, but I do need attended transfers, even if the ringback is always with G711. SIP transfers in general are relatively new to me, so I very well may be asking for my cake, eating it, and then wanting the bakery to deliver daily to my office for free. Forgive me if that's the case. - Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/3fb76f51/attachment.html From steveayre at gmail.com Mon Jan 14 23:34:20 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 14 Jan 2013 20:34:20 +0000 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: References: <439D90B8-16C2-457C-82EB-524D179A2486@jerris.com> Message-ID: "All codecs are fully indemnified; no additional licensing is required for their use." Source: http://sangoma.com/products/media_processing/voice_transcoding_boards/d150.html On 14 January 2013 19:58, Kristian Kielhofner wrote: > IIRC Sangoma doesn't provide licensing for AMR and AMR-WB (just the > implementation). > > AFAIK mod_sangoma_codec is no more legal than libamr. From steveayre at gmail.com Mon Jan 14 23:39:17 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 14 Jan 2013 20:39:17 +0000 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: References: <439D90B8-16C2-457C-82EB-524D179A2486@jerris.com> Message-ID: Ah, damn you're right Kristian! The data sheet http://sangoma.com/assets/docs/datasheets/en/d100.pdf contains this footnode that is omitted from the web page: "Except for AMR and AMR-WB. Integrators must clear patent license with VoiceAge Corp." :( In which case anyone obtaining licenses from VoiceAge Corp should be free to use them with any implementation... I guess this is due to the same patent pool mess Mike mentioned? -Steve On 14 January 2013 20:34, Steven Ayre wrote: > "All codecs are fully indemnified; no additional licensing is required > for their use." > Source: http://sangoma.com/products/media_processing/voice_transcoding_boards/d150.html > > > On 14 January 2013 19:58, Kristian Kielhofner wrote: >> IIRC Sangoma doesn't provide licensing for AMR and AMR-WB (just the >> implementation). >> >> AFAIK mod_sangoma_codec is no more legal than libamr. From kris at kriskinc.com Mon Jan 14 23:57:39 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 14 Jan 2013 15:57:39 -0500 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: References: <439D90B8-16C2-457C-82EB-524D179A2486@jerris.com> Message-ID: My understanding is that AMR licensing is even more ridiculous than G729. On Mon, Jan 14, 2013 at 3:39 PM, Steven Ayre wrote: > Ah, damn you're right Kristian! > > The data sheet > http://sangoma.com/assets/docs/datasheets/en/d100.pdf > contains this footnode that is omitted from the web page: > > "Except for AMR and AMR-WB. Integrators must clear patent license with > VoiceAge Corp." > > :( > > In which case anyone obtaining licenses from VoiceAge Corp should be > free to use them with any implementation... > > I guess this is due to the same patent pool mess Mike mentioned? > > -Steve > > > On 14 January 2013 20:34, Steven Ayre wrote: >> "All codecs are fully indemnified; no additional licensing is required >> for their use." >> Source: http://sangoma.com/products/media_processing/voice_transcoding_boards/d150.html >> >> >> On 14 January 2013 19:58, Kristian Kielhofner wrote: >>> IIRC Sangoma doesn't provide licensing for AMR and AMR-WB (just the >>> implementation). >>> >>> AFAIK mod_sangoma_codec is no more legal than libamr. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From krice at freeswitch.org Tue Jan 15 00:10:18 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 14 Jan 2013 15:10:18 -0600 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: Message-ID: Actually its crazy rediculous... Companies like Sangoma cant even license it. That requirement for integrators isnt sangoma's, its VoiceAge's requirement On 1/14/13 2:57 PM, "Kristian Kielhofner" wrote: > My understanding is that AMR licensing is even more ridiculous than G729. > > On Mon, Jan 14, 2013 at 3:39 PM, Steven Ayre wrote: >> Ah, damn you're right Kristian! >> >> The data sheet >> http://sangoma.com/assets/docs/datasheets/en/d100.pdf >> contains this footnode that is omitted from the web page: >> >> "Except for AMR and AMR-WB. Integrators must clear patent license with >> VoiceAge Corp." >> >> :( >> >> In which case anyone obtaining licenses from VoiceAge Corp should be >> free to use them with any implementation... >> >> I guess this is due to the same patent pool mess Mike mentioned? >> >> -Steve >> >> >> On 14 January 2013 20:34, Steven Ayre wrote: >>> "All codecs are fully indemnified; no additional licensing is required >>> for their use." >>> Source: >>> http://sangoma.com/products/media_processing/voice_transcoding_boards/d150.h >>> tml >>> >>> >>> On 14 January 2013 19:58, Kristian Kielhofner wrote: >>>> IIRC Sangoma doesn't provide licensing for AMR and AMR-WB (just the >>>> implementation). >>>> >>>> AFAIK mod_sangoma_codec is no more legal than libamr. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From msc at freeswitch.org Tue Jan 15 00:24:57 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jan 2013 13:24:57 -0800 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: References: Message-ID: Bummer. Now we need to start collecting some appropriate backronyms for "AMR" and "AMR-WB". -MC On Mon, Jan 14, 2013 at 1:10 PM, Ken Rice wrote: > Actually its crazy rediculous... Companies like Sangoma cant even license > it. That requirement for integrators isnt sangoma's, its VoiceAge's > requirement > > > > On 1/14/13 2:57 PM, "Kristian Kielhofner" wrote: > > > My understanding is that AMR licensing is even more ridiculous than G729. > > > > On Mon, Jan 14, 2013 at 3:39 PM, Steven Ayre > wrote: > >> Ah, damn you're right Kristian! > >> > >> The data sheet > >> http://sangoma.com/assets/docs/datasheets/en/d100.pdf > >> contains this footnode that is omitted from the web page: > >> > >> "Except for AMR and AMR-WB. Integrators must clear patent license with > >> VoiceAge Corp." > >> > >> :( > >> > >> In which case anyone obtaining licenses from VoiceAge Corp should be > >> free to use them with any implementation... > >> > >> I guess this is due to the same patent pool mess Mike mentioned? > >> > >> -Steve > >> > >> > >> On 14 January 2013 20:34, Steven Ayre wrote: > >>> "All codecs are fully indemnified; no additional licensing is required > >>> for their use." > >>> Source: > >>> > http://sangoma.com/products/media_processing/voice_transcoding_boards/d150.h > >>> tml > >>> > >>> > >>> On 14 January 2013 19:58, Kristian Kielhofner > wrote: > >>>> IIRC Sangoma doesn't provide licensing for AMR and AMR-WB (just the > >>>> implementation). > >>>> > >>>> AFAIK mod_sangoma_codec is no more legal than libamr. > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/5c3a464d/attachment.html From mike at jerris.com Tue Jan 15 00:55:33 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 14 Jan 2013 16:55:33 -0500 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: References: <439D90B8-16C2-457C-82EB-524D179A2486@jerris.com> Message-ID: I would NOT assume that the VoiceAge pool is complete. On Jan 14, 2013, at 3:39 PM, Steven Ayre wrote: > Ah, damn you're right Kristian! > > The data sheet > http://sangoma.com/assets/docs/datasheets/en/d100.pdf > contains this footnode that is omitted from the web page: > > "Except for AMR and AMR-WB. Integrators must clear patent license with > VoiceAge Corp." > > :( > > In which case anyone obtaining licenses from VoiceAge Corp should be > free to use them with any implementation... > > I guess this is due to the same patent pool mess Mike mentioned? > > -Steve > > > On 14 January 2013 20:34, Steven Ayre wrote: >> "All codecs are fully indemnified; no additional licensing is required >> for their use." >> Source: http://sangoma.com/products/media_processing/voice_transcoding_boards/d150.html >> >> >> On 14 January 2013 19:58, Kristian Kielhofner wrote: >>> IIRC Sangoma doesn't provide licensing for AMR and AMR-WB (just the >>> implementation). >>> >>> AFAIK mod_sangoma_codec is no more legal than libamr. From brent.paddon at gmail.com Tue Jan 15 01:02:33 2013 From: brent.paddon at gmail.com (Brent Paddon) Date: Tue, 15 Jan 2013 08:02:33 +1000 Subject: [Freeswitch-users] Delay Round Two...FIGHT! In-Reply-To: References: <50F42891.7090000@quentustech.com> Message-ID: Figure out how to use Wireshark and a SPAN (mirror) port on your switch(es) and things will become a lot clearer for you. You'll be able to see what is actually happening on the network instead of guessing. Brent On Tue, Jan 15, 2013 at 2:32 AM, Blake Priddy wrote: > Well our server has 6 cores. William, we are running stable on debian. Any > other brain busters? I need as many possibilites of issues as possible. > Roger, we have gig across the network. But I will give it a shot just to > eliminate that possibility. > > > On Mon, Jan 14, 2013 at 10:10 AM, Roger Castaldo > wrote: > >> Another option, to at least eliminate issues, assuming it happens on all >> calls is to connect a phone either directly to the server, ie spare nic >> port or as close as possible, ie same switch and perform an external call >> to that phone to see if the delay occurs. This would at least allow you to >> eliminate a network latency between the server and the secretaries phone >> which would cause delays in audio as well since freeswitch cannot send >> audio if the audio hasn't arrived to it yet. Just a thought on trying to >> eliminate the possible. >> >> >> On Mon, Jan 14, 2013 at 10:47 AM, William King < >> william.king at quentustech.com> wrote: >> >>> Do you happen to be running the stable branch? If so then I believe the >>> issue you are hitting was already fixed in master. If you have a chance >>> to test, see if the length of the audio delay is about the same amount >>> of time as the secretaries phone was ringing. If that is the case, then >>> it should already be fixed in master. >>> >>> William King >>> Senior Engineer >>> Quentus Technologies, INC >>> 1037 NE 65th St Suite 273 >>> Seattle, WA 98115 >>> Main: (877) 211-9337 >>> Office: (206) 388-4772 >>> Cell: (253) 686-5518 >>> william.king at quentustech.com >>> >>> On 01/14/2013 06:45 AM, Blake Priddy wrote: >>> > Ok everyone, hope your new years went wonderful! So here is the >>> > situation again. THE DELAY IS BACK! I have wiped FS and installed the >>> > latest git head and the delay is still present. I want to eliminate all >>> > FS options that could fix it before I call AT&T. Any help is >>> > appreciated! Person from outside the school calls, secretary answers, >>> > secretary hears they person saying hello, secretary says hello over and >>> > over again for 10 seconds. Then the B-Leg audio finally is heard by the >>> > A-LEG. Note* The B-Leg (secretary) can hear the A-leg talking the whole >>> > time. Just the A-Leg cannot hear the (secretary) B-Leg until about 7-13 >>> > seconds has passed. I see nothing in the FS log after the audio from >>> the >>> > B-Leg finally traverses to the A-Leg. Please help our school fellow FS >>> > users! >>> > >>> > LONG LIVE FS!!! :) >>> > >>> > -- >>> > >>> > *Blakelund Priddy* >>> > Network Systems Engineer >>> > Bryant Public School District >>> > Bryant, Arkansas 72022 >>> > http://www.bryantschools.org >>> > p 501-653-5038 >>> > f 501-847-5656 >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > > *Blakelund Priddy* > > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/0804aee6/attachment-0001.html From schoch+freeswitch.org at xwin32.com Tue Jan 15 01:13:19 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 14 Jan 2013 14:13:19 -0800 Subject: [Freeswitch-users] Multiple Network Interfaces In-Reply-To: References: Message-ID: On Mon, Jan 14, 2013 at 11:36 AM, Avi Marcus wrote: > Just to clarify: do you have one sofia profile per IP:port? > I think I'm starting to get a hint of what the problem is. :-) Apparently, both the "internal" and "external" sofia profiles use $local_ip_v4, which is set to 192.168.4.1. I'm thinking my best plan is to replace the "internal" profile with "phones", "lan" and "wan" profiles. So, the answer to your question is: "No, I just have the default sofia profiles." But now I believe I have learned that a single profile cannot span multiple IP addresses, correct? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/74b6cc8a/attachment.html From freeswitch at nullinfinity.org Tue Jan 15 00:59:33 2013 From: freeswitch at nullinfinity.org (Johan Brannlund) Date: Mon, 14 Jan 2013 16:59:33 -0500 Subject: [Freeswitch-users] Voicemail and forwarding MWI Message-ID: <20130114165933.29c9a361@mars> Hello. I recently installed Freeswitch for use as a home PBX. I was planning on using my VoIP provider's voicemail system, but I'm not sure if that's possible. In particular, I don't know how to get MWI working in such a configuration. Is there a way for Freeswitch to forward the SIP NOTIFY MWI messages from my provider to a certain extension? In other words, I am trying to do exactly the same thing that Gabriel Burca asked about a year ago (see http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078703.html ). Any help would be much appreciated. From msc at freeswitch.org Tue Jan 15 01:29:26 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jan 2013 14:29:26 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Greetings! It has been a light week when it comes to news items. However, the FreeSWITCH team has been working hard on bug fixes and various projects. More information on those will come out once the details are available. Last week we had a community discussion on the weekly conference call, mostly discussing the ways that we can all help out with getting Jira's handled. Although we did not formally adopt the "Jira karma" idea we decided that we (as a project and community) would do our best to reward those who help out with bug marshaling, verifying, documenting, etc. Every little bit helps. To that end this weekwe will be spending some time as a group going over a few selected bugs and as a group trying our best to figure what's going on. Ken and Brian will supply us with a few bugs to tackle. Oh, and if you use Cisco phones, please be very careful ! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/1681fbaf/attachment.html From avi at avimarcus.net Tue Jan 15 01:32:55 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 15 Jan 2013 00:32:55 +0200 Subject: [Freeswitch-users] Multiple Network Interfaces In-Reply-To: References: Message-ID: Correct - one profile per ip:port. -Avi On Tue, Jan 15, 2013 at 12:13 AM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Mon, Jan 14, 2013 at 11:36 AM, Avi Marcus wrote: > >> Just to clarify: do you have one sofia profile per IP:port? >> > > I think I'm starting to get a hint of what the problem is. :-) > > Apparently, both the "internal" and "external" sofia profiles use > $local_ip_v4, which is set to 192.168.4.1. I'm thinking my best plan is to > replace the "internal" profile with "phones", "lan" and "wan" profiles. > > So, the answer to your question is: "No, I just have the default sofia > profiles." But now I believe I have learned that a single profile cannot > span multiple IP addresses, correct? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/45fa9afc/attachment.html From anthony.minessale at gmail.com Tue Jan 15 01:33:40 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Jan 2013 16:33:40 -0600 Subject: [Freeswitch-users] Delay Round Two...FIGHT! In-Reply-To: References: <50F42891.7090000@quentustech.com> Message-ID: Words like delay can lead to a lot of confusion because there are so many ways the word is used. The delay problem William is talking about is a permanent delay that was a brief bug we had where the audio streams were always out of sync. The problem described sounds more like "setup delay" which is: The audio is in sync but its not bi-directional for a long period of time. This sounds to me like one of the many Sonus bugs we have fought where passing rtp streams to a Sonus would cause it to restart its jitter buffer and eat up audio. What you should do follow these steps: 1) Install GIT HEAD (not stable branch) because the stable branch cannot be modified and the code is too old to diagnose problems on. 2) Reproduce the problem and get traces / log caps. (if you cannot reproduce it on demand its likely an environmental issue since most delay bugs that are our fault are consistent) 3) Report them to JIRA not to this mailing list. Problems are impossible to track over email... That's why we have an issue tracker. Re-reading this : I see you mentioned ATT so its almost for sure the Sonus bug. Try this: edit your sofia profile and set the param rtp-rewrite-timestamps to true My suspicion: The ATT calls you, then it bridges over to FS that maybe sends some data like playing a "please hold" sound or enters a queue something. Then bridges to a phone. By default we preserve timestamps on RTP so when we send the stamps from the phone it confuses it and makes it go insane. This param will rewrite all timestamps so it always sees them coming from a generated base rather than passing them. BTW: "thousands of calls per second" ... no ... That's why we don't touch the subject of load testing.... On Mon, Jan 14, 2013 at 4:02 PM, Brent Paddon wrote: > Figure out how to use Wireshark and a SPAN (mirror) port on your > switch(es) and things will become a lot clearer for you. You'll be able to > see what is actually happening on the network instead of guessing. > > Brent > > > On Tue, Jan 15, 2013 at 2:32 AM, Blake Priddy wrote: > >> Well our server has 6 cores. William, we are running stable on debian. >> Any other brain busters? I need as many possibilites of issues as possible. >> Roger, we have gig across the network. But I will give it a shot just to >> eliminate that possibility. >> >> >> On Mon, Jan 14, 2013 at 10:10 AM, Roger Castaldo < >> roger.castaldo at gmail.com> wrote: >> >>> Another option, to at least eliminate issues, assuming it happens on all >>> calls is to connect a phone either directly to the server, ie spare nic >>> port or as close as possible, ie same switch and perform an external call >>> to that phone to see if the delay occurs. This would at least allow you to >>> eliminate a network latency between the server and the secretaries phone >>> which would cause delays in audio as well since freeswitch cannot send >>> audio if the audio hasn't arrived to it yet. Just a thought on trying to >>> eliminate the possible. >>> >>> >>> On Mon, Jan 14, 2013 at 10:47 AM, William King < >>> william.king at quentustech.com> wrote: >>> >>>> Do you happen to be running the stable branch? If so then I believe the >>>> issue you are hitting was already fixed in master. If you have a chance >>>> to test, see if the length of the audio delay is about the same amount >>>> of time as the secretaries phone was ringing. If that is the case, then >>>> it should already be fixed in master. >>>> >>>> William King >>>> Senior Engineer >>>> Quentus Technologies, INC >>>> 1037 NE 65th St Suite 273 >>>> Seattle, WA 98115 >>>> Main: (877) 211-9337 >>>> Office: (206) 388-4772 >>>> Cell: (253) 686-5518 >>>> william.king at quentustech.com >>>> >>>> On 01/14/2013 06:45 AM, Blake Priddy wrote: >>>> > Ok everyone, hope your new years went wonderful! So here is the >>>> > situation again. THE DELAY IS BACK! I have wiped FS and installed the >>>> > latest git head and the delay is still present. I want to eliminate >>>> all >>>> > FS options that could fix it before I call AT&T. Any help is >>>> > appreciated! Person from outside the school calls, secretary answers, >>>> > secretary hears they person saying hello, secretary says hello over >>>> and >>>> > over again for 10 seconds. Then the B-Leg audio finally is heard by >>>> the >>>> > A-LEG. Note* The B-Leg (secretary) can hear the A-leg talking the >>>> whole >>>> > time. Just the A-Leg cannot hear the (secretary) B-Leg until about >>>> 7-13 >>>> > seconds has passed. I see nothing in the FS log after the audio from >>>> the >>>> > B-Leg finally traverses to the A-Leg. Please help our school fellow FS >>>> > users! >>>> > >>>> > LONG LIVE FS!!! :) >>>> > >>>> > -- >>>> > >>>> > *Blakelund Priddy* >>>> > Network Systems Engineer >>>> > Bryant Public School District >>>> > Bryant, Arkansas 72022 >>>> > http://www.bryantschools.org >>>> > p 501-653-5038 >>>> > f 501-847-5656 >>>> > >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> >> *Blakelund Priddy* >> >> Network Systems Engineer >> Bryant Public School District >> Bryant, Arkansas 72022 >> http://www.bryantschools.org >> p 501-653-5038 >> f 501-847-5656 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/42ccfdc6/attachment-0001.html From steveayre at gmail.com Tue Jan 15 01:42:07 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 14 Jan 2013 22:42:07 +0000 Subject: [Freeswitch-users] Multiple Network Interfaces In-Reply-To: References: Message-ID: Yep, the default configs are only intended as an example. For a multi-homed server you *must* configure at least one profile for each IP you want FS to listen on. It's not possible for FS to listen on multiple IPs as the SIP packets must include the IP to reach FS on for SIP responses (eg Contact header) and RTP media (SDP). That IP is obviously different depending on which IP you're connecting to. -Steve On 14 January 2013 22:13, Steven Schoch wrote: > On Mon, Jan 14, 2013 at 11:36 AM, Avi Marcus wrote: >> >> Just to clarify: do you have one sofia profile per IP:port? > > > I think I'm starting to get a hint of what the problem is. :-) > > Apparently, both the "internal" and "external" sofia profiles use > $local_ip_v4, which is set to 192.168.4.1. I'm thinking my best plan is to > replace the "internal" profile with "phones", "lan" and "wan" profiles. > > So, the answer to your question is: "No, I just have the default sofia > profiles." But now I believe I have learned that a single profile cannot > span multiple IP addresses, correct? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveayre at gmail.com Tue Jan 15 01:43:39 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 14 Jan 2013 22:43:39 +0000 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: References: Message-ID: In that case the same issue would go for FreeSWITCH too...? I guess that's why mod_amr is still passthrough-only despite all the fantastic work that was put into mod_com_g729, otherwise it presumable would have been very easy to have the same license server code reused for AMR. -Steve On 14 January 2013 21:10, Ken Rice wrote: > Actually its crazy rediculous... Companies like Sangoma cant even license > it. That requirement for integrators isnt sangoma's, its VoiceAge's > requirement > > > > On 1/14/13 2:57 PM, "Kristian Kielhofner" wrote: > >> My understanding is that AMR licensing is even more ridiculous than G729. >> >> On Mon, Jan 14, 2013 at 3:39 PM, Steven Ayre wrote: >>> Ah, damn you're right Kristian! >>> >>> The data sheet >>> http://sangoma.com/assets/docs/datasheets/en/d100.pdf >>> contains this footnode that is omitted from the web page: >>> >>> "Except for AMR and AMR-WB. Integrators must clear patent license with >>> VoiceAge Corp." >>> >>> :( >>> >>> In which case anyone obtaining licenses from VoiceAge Corp should be >>> free to use them with any implementation... >>> >>> I guess this is due to the same patent pool mess Mike mentioned? >>> >>> -Steve >>> >>> >>> On 14 January 2013 20:34, Steven Ayre wrote: >>>> "All codecs are fully indemnified; no additional licensing is required >>>> for their use." >>>> Source: >>>> http://sangoma.com/products/media_processing/voice_transcoding_boards/d150.h >>>> tml >>>> >>>> >>>> On 14 January 2013 19:58, Kristian Kielhofner wrote: >>>>> IIRC Sangoma doesn't provide licensing for AMR and AMR-WB (just the >>>>> implementation). >>>>> >>>>> AFAIK mod_sangoma_codec is no more legal than libamr. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Tue Jan 15 02:18:13 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jan 2013 15:18:13 -0800 Subject: [Freeswitch-users] Early media handling on outbound call In-Reply-To: References: Message-ID: So you're saying that the provider is sending inband audio in early media saying that the call is not answering, and then on the same call you hear ringing? -MC On Sun, Jan 13, 2013 at 6:28 AM, Hynek Cihlar wrote: > Scenario: > A call is originated to a mobile network. Mobile phone starts ringing and > is hung up. The mobile phone leg is pre-answered and early media is > returned saying that the caller is not answering. The message is playbed > back twice, than silence for cca two seconds and then ringing tone with the > actual phone starting to ring as well. The phone is hung up, early media > played back, etc... > > The freeswitch console command used to originate the call is as follows: > originate sofia/internal/1%terminal_domain > &bridge({origination_caller_id_number= number>}sofia/gateway/axfone/) > > My question is. Is this expected behavior? Should not the destination > server end the call after early media is played back? Or is it expected > from me to detect a silence and end the call? If so, what would be the best > approach to solve it, wait_for_silence? > > Thanks for comments! > Hynek > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/461de363/attachment.html From steveu at coppice.org Tue Jan 15 02:35:33 2013 From: steveu at coppice.org (Steve Underwood) Date: Tue, 15 Jan 2013 07:35:33 +0800 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: References: <439D90B8-16C2-457C-82EB-524D179A2486@jerris.com> Message-ID: <50F49645.8080304@coppice.org> There isn't really anything ridiculous about the licencing of G.729. It may be annoying, but its pretty much the only encumbered codec where there is a reasonably well defined path for fairly small scale users to licence all the relevant patents (even if one or two do crawl out of the woodwork late in the day via letters from lawyers). G.723.1, AMR, AMR-WB, EVRC, GSM-HR, GSM-EFR, etc., are all a PITA to licence. Regards, Steve On 01/15/2013 04:57 AM, Kristian Kielhofner wrote: > My understanding is that AMR licensing is even more ridiculous than G729. > > On Mon, Jan 14, 2013 at 3:39 PM, Steven Ayre wrote: >> Ah, damn you're right Kristian! >> >> The data sheet >> http://sangoma.com/assets/docs/datasheets/en/d100.pdf >> contains this footnode that is omitted from the web page: >> >> "Except for AMR and AMR-WB. Integrators must clear patent license with >> VoiceAge Corp." >> >> :( >> >> In which case anyone obtaining licenses from VoiceAge Corp should be >> free to use them with any implementation... >> >> I guess this is due to the same patent pool mess Mike mentioned? >> >> -Steve >> >> >> On 14 January 2013 20:34, Steven Ayre wrote: >>> "All codecs are fully indemnified; no additional licensing is required >>> for their use." >>> Source: http://sangoma.com/products/media_processing/voice_transcoding_boards/d150.html >>> >>> >>> On 14 January 2013 19:58, Kristian Kielhofner wrote: >>>> IIRC Sangoma doesn't provide licensing for AMR and AMR-WB (just the >>>> implementation). >>>> >>>> AFAIK mod_sangoma_codec is no more legal than libamr. >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > From lists at kavun.ch Tue Jan 15 02:36:51 2013 From: lists at kavun.ch (Emrah) Date: Mon, 14 Jan 2013 18:36:51 -0500 Subject: [Freeswitch-users] Delay Round Two...FIGHT! In-Reply-To: References: Message-ID: <35D2B339-B2C0-456F-94D1-6DA5921B6E11@kavun.ch> Hi there, I had this issue recently. I think it appeared when I added bridge_early_media=true before setting up the call. This was 2 versions prior to the current today. I don't know if the bridge_early_media is a coincidence and if it was the version I used that had an issue. But I am very thankful that the issue is now past? Good luck and all the best, Emrah Wiki page: http://wiki.freeswitch.org/wiki/Variable_bridge_early_media On Jan 14, 2013, at 9:45 AM, Blake Priddy wrote: > Ok everyone, hope your new years went wonderful! So here is the situation again. THE DELAY IS BACK! I have wiped FS and installed the latest git head and the delay is still present. I want to eliminate all FS options that could fix it before I call AT&T. Any help is appreciated! Person from outside the school calls, secretary answers, secretary hears they person saying hello, secretary says hello over and over again for 10 seconds. Then the B-Leg audio finally is heard by the A-LEG. Note* The B-Leg (secretary) can hear the A-leg talking the whole time. Just the A-Leg cannot hear the (secretary) B-Leg until about 7-13 seconds has passed. I see nothing in the FS log after the audio from the B-Leg finally traverses to the A-Leg. Please help our school fellow FS users! > > LONG LIVE FS!!! :) > > -- > > Blakelund Priddy > Network Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lists at kavun.ch Tue Jan 15 02:49:21 2013 From: lists at kavun.ch (Emrah) Date: Mon, 14 Jan 2013 18:49:21 -0500 Subject: [Freeswitch-users] Is Asterisk snappier than FreeSWITCH to set up calls? Message-ID: Ever since I converted to FS, I consistently felt a little delay in setting up my calls. I thought it was just me being too picky or that the roundtrip required for the digest auth was the cause? Since I now learned that Digest is also required by Asterisk, I am out of ideas? I tested both SIP servers in parallel and captured a trace. On my Asterisk, the extension 114 dials an IAX peer. The kind of thing that should add more delay? On my FS box, it's milliwatt, executed immediately. Asterisk: http://pastebin.freeswitch.org/20445 FS: http://pastebin.freeswitch.org/20444 Why is Asterisk snappier? I set all log levels to 0 on my FS box? Thanks for any ideas. Best, Emrah From kris at kriskinc.com Tue Jan 15 02:55:53 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 14 Jan 2013 18:55:53 -0500 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: <50F49645.8080304@coppice.org> References: <439D90B8-16C2-457C-82EB-524D179A2486@jerris.com> <50F49645.8080304@coppice.org> Message-ID: Compared to free codecs any amount of licensing is ridiculous by comparison :). On Mon, Jan 14, 2013 at 6:35 PM, Steve Underwood wrote: > There isn't really anything ridiculous about the licencing of G.729. It > may be annoying, but its pretty much the only encumbered codec where > there is a reasonably well defined path for fairly small scale users to > licence all the relevant patents (even if one or two do crawl out of the > woodwork late in the day via letters from lawyers). G.723.1, AMR, > AMR-WB, EVRC, GSM-HR, GSM-EFR, etc., are all a PITA to licence. > > Regards, > Steve > > On 01/15/2013 04:57 AM, Kristian Kielhofner wrote: >> My understanding is that AMR licensing is even more ridiculous than G729. >> >> On Mon, Jan 14, 2013 at 3:39 PM, Steven Ayre wrote: >>> Ah, damn you're right Kristian! >>> >>> The data sheet >>> http://sangoma.com/assets/docs/datasheets/en/d100.pdf >>> contains this footnode that is omitted from the web page: >>> >>> "Except for AMR and AMR-WB. Integrators must clear patent license with >>> VoiceAge Corp." >>> >>> :( >>> >>> In which case anyone obtaining licenses from VoiceAge Corp should be >>> free to use them with any implementation... >>> >>> I guess this is due to the same patent pool mess Mike mentioned? >>> >>> -Steve >>> >>> >>> On 14 January 2013 20:34, Steven Ayre wrote: >>>> "All codecs are fully indemnified; no additional licensing is required >>>> for their use." >>>> Source: http://sangoma.com/products/media_processing/voice_transcoding_boards/d150.html >>>> >>>> >>>> On 14 January 2013 19:58, Kristian Kielhofner wrote: >>>>> IIRC Sangoma doesn't provide licensing for AMR and AMR-WB (just the >>>>> implementation). >>>>> >>>>> AFAIK mod_sangoma_codec is no more legal than libamr. >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From kris at kriskinc.com Tue Jan 15 02:59:37 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 14 Jan 2013 18:59:37 -0500 Subject: [Freeswitch-users] Is Asterisk snappier than FreeSWITCH to set up calls? In-Reply-To: References: Message-ID: You haven't provided a console log but it's possible you're experiencing this: http://jira.freeswitch.org/browse/FS-4910 On Mon, Jan 14, 2013 at 6:49 PM, Emrah wrote: > Ever since I converted to FS, I consistently felt a little delay in setting up my calls. I thought it was just me being too picky or that the roundtrip required for the digest auth was the cause? > Since I now learned that Digest is also required by Asterisk, I am out of ideas? > > I tested both SIP servers in parallel and captured a trace. > On my Asterisk, the extension 114 dials an IAX peer. The kind of thing that should add more delay? > On my FS box, it's milliwatt, executed immediately. > > Asterisk: http://pastebin.freeswitch.org/20445 > FS: http://pastebin.freeswitch.org/20444 > > Why is Asterisk snappier? > > I set all log levels to 0 on my FS box? > > Thanks for any ideas. > > Best, > Emrah > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From msc at freeswitch.org Tue Jan 15 03:10:52 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jan 2013 16:10:52 -0800 Subject: [Freeswitch-users] Is Asterisk snappier than FreeSWITCH to set up calls? In-Reply-To: References: Message-ID: Assuming that both servers are absolutely identical and running under the exact same amount of load, i.e. that "server resources" are not a factor, then it looks like Sofia's dialog is slightly chattier and a fraction of a second slower than the equivalent dialog on Asterisk. Keep in mind, though, that this is for a single call. Send 20, 30, 40 CPS or more for an extended period of time and see what the call setup times are. I suspect one of your servers will scale a lot more gracefully than the other... -MC On Mon, Jan 14, 2013 at 3:49 PM, Emrah wrote: > Ever since I converted to FS, I consistently felt a little delay in > setting up my calls. I thought it was just me being too picky or that the > roundtrip required for the digest auth was the cause? > Since I now learned that Digest is also required by Asterisk, I am out of > ideas? > > I tested both SIP servers in parallel and captured a trace. > On my Asterisk, the extension 114 dials an IAX peer. The kind of thing > that should add more delay? > On my FS box, it's milliwatt, executed immediately. > > Asterisk: http://pastebin.freeswitch.org/20445 > FS: http://pastebin.freeswitch.org/20444 > > Why is Asterisk snappier? > > I set all log levels to 0 on my FS box? > > Thanks for any ideas. > > Best, > Emrah > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/36d6525e/attachment.html From msc at freeswitch.org Tue Jan 15 03:16:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jan 2013 16:16:30 -0800 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: References: <23602598.3790.1358184950672.JavaMail.javamailuser@localhost> <15959415.3795.1358184967040.JavaMail.javamailuser@localhost> Message-ID: On Mon, Jan 14, 2013 at 10:34 AM, Dmitry Sytchev wrote: > We started development of web-ui rather as programming exercise in JS, > twitter bootstrap and perl Mojolicious :) > Perl mojolicious is very cool, btw. -MC > > Yes, engine itself does all stuff we need to analyze calls, it stores data > in db, can be set up to use multiple sensors and store their trace data in > same DB. Obviously, pcaps and other large raw media data is stored locally > on sensor machines. > > We use it to do full monitoring of all legs of all VoIP calls in our > network. > > > > 2013/1/14 Raimund Sacherer > >> Thank you Dmitry, >> >> are you saying you only use the opensource capture part from voip-monitor >> on all your 10 FS servers and you do not use their webinterface at all? If >> I understood this correctly, this opensource does all the capturing and it >> calculates MOS and Loss scores and stores it in a central MySQL database? >> >> That sound's great, I thought the MOS and Loss calculations are in their >> proprietery part, but if it's included in the opensource part it could be >> interesting to create your own web-visualization. On the other hand their >> licence seems to be fairly cheap, some 480 euros ... but I am not sure if >> this is per server you monitor, or per server you deploy the web-interface >> to. >> >> best, >> Rai >> >> ------------------------------ >> *From: *"Dmitry Sytchev" >> >> *To: *"FreeSWITCH Users Help" >> *Sent: *Viernes, 11 de Enero 2013 14:35:33 >> *Subject: *Re: [Freeswitch-users] Bounty for - FS-4268 Sip trace with >> filter for IP or extension >> >> >> We use VoipMonitor open-source part, with homebrew custom WEB-UI. This is >> a really cool software, it continuously evolves and captures pretty well >> even large amounts of VoiP traffic. >> >> I highly recommend this software, since it doesn't only captures >> signaling traffic, but also can capture media/rtcp, automatically convert >> captured calls to wav files or store raw pcaps for further processing. it >> also can analyze MOS/Loss scores and many many other different parameters >> and store them in database per call. >> >> We use single instance to store data for investigation of calls with >> problems on more than 10 servers. >> >> >> >> >> 2013/1/11 Raimund Sacherer >> >>> Does anyone have any experience with voip monitor? >>> http://www.voipmonitor.org/ >>> >>> We will start evaluating it in our environment as we are switching more >>> and more to SIP providers, therefore I would love to have better >>> Monitoring. >>> >>> What voipmonitor says is that they are able to filter the calls in >>> realtime with low overhead an if call quality drops, they can alert you, >>> and they seem to have a rather nice web interface and they also are used by >>> others to store the RTP's ... >>> >>> What seems to be especially nice is that their capture daemon (as they >>> state it) adds very little overhead to a system, so depending on call >>> quantity they say it's neglect-able. >>> >>> Would love to hear some comparisions between this and HOMER if some can >>> provide it. >>> >>> Thank you, best >>> Ray >>> >>> ------------------------------ >>> *From: *"Cal Leeming [Simplicity Media Ltd]" < >>> cal.leeming at simplicitymedialtd.co.uk> >>> *To: *"FreeSWITCH Users Help" >>> *Sent: *Domingo, 6 de Enero 2013 21:01:24 >>> *Subject: *[Freeswitch-users] Bounty for - FS-4268 Sip trace with >>> filter for IP or extension >>> >>> >>> Could a core dev speculate on how much the expected bounty would be for >>> this? >>> >>> Thanks >>> >>> Cal >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> -- >>> [image: Logitravel.com] Raimund Sacherer >>> *Sistemas*Agencia de Viajes Online >>> www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) >>> Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca >>> Tel 902 366 847 | Fax 971 213 495 S?guenos en:[image: Facebook de >>> Logitravel] [image: Twitter de >>> Logitravel] [image: Blog de Logitravel] [image: >>> Logitravel en Youtube] [image: >>> Logitravel en Foursquare] Descarga >>> nuestras *aplicaciones para m?vil* [image: Logitravel.com]Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene >>> informaci?n de car?cter confidencial exclusivamente dirigida a su >>> destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a >>> terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de >>> haber recibido este correo electr?nico por error, se ruega notif?quese >>> inmediatamente esta circunstancia mediante reenv?o a la direcci?n >>> electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus >>> datos forman o formar?n parte de un fichero registrado como CLIENTES con >>> n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de >>> Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio >>> Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. >>> Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a >>> estos datos que deber? ejercer mediante escrito a la direcci?n >>> anteriormente citada. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Best regards, >> >> Dmitry Sytchev, >> IT Engineer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Best regards, > > Dmitry Sytchev, > IT Engineer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/2f06cadb/attachment-0001.html From schoch+freeswitch.org at xwin32.com Tue Jan 15 03:23:02 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 14 Jan 2013 16:23:02 -0800 Subject: [Freeswitch-users] FAX problems Message-ID: Thank you guys, I just sorted out the multiple sofia profiles and have FS accepting SIP registrations on our LAN. I plugged a phone into the LAN and it works. Now I'm trying to get this VoIP Plug-In for Microsoft Fax to work, that I downloaded from www.faxback.com. I configured it with a directory number I created, the dialplan seems to work, but it fails when sending the test FAX. The FAX number I'm calling is a POTS line connected to a Fax modem, which is answered by Hylafax. This works and is used everyday. The faxback software is installed on a Windows 7 machine, and configured by putting in a proxy address of 192.168.1.17, a username and password, and address of user at domain. The test call rings the fax modem, it answers, but then after some handshaking, it hangs up and the window says the call was hung up. Hylafax says: "No answer (T.30 T1 timeout)" -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/c2a810b7/attachment.html From kris at kriskinc.com Tue Jan 15 04:04:31 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 14 Jan 2013 20:04:31 -0500 Subject: [Freeswitch-users] Is Asterisk snappier than FreeSWITCH to set up calls? In-Reply-To: References: Message-ID: Michael, Our experience with that issue is very real. I provided details in my comment but call setup time jumps to almost 2 seconds: http://jira.freeswitch.org/browse/FS-4910?focusedCommentId=38458&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-38458 On Mon, Jan 14, 2013 at 7:10 PM, Michael Collins wrote: > Assuming that both servers are absolutely identical and running under the > exact same amount of load, i.e. that "server resources" are not a factor, > then it looks like Sofia's dialog is slightly chattier and a fraction of a > second slower than the equivalent dialog on Asterisk. Keep in mind, though, > that this is for a single call. Send 20, 30, 40 CPS or more for an extended > period of time and see what the call setup times are. I suspect one of your > servers will scale a lot more gracefully than the other... > > -MC > > On Mon, Jan 14, 2013 at 3:49 PM, Emrah wrote: >> >> Ever since I converted to FS, I consistently felt a little delay in >> setting up my calls. I thought it was just me being too picky or that the >> roundtrip required for the digest auth was the cause? >> Since I now learned that Digest is also required by Asterisk, I am out of >> ideas? >> >> I tested both SIP servers in parallel and captured a trace. >> On my Asterisk, the extension 114 dials an IAX peer. The kind of thing >> that should add more delay? >> On my FS box, it's milliwatt, executed immediately. >> >> Asterisk: http://pastebin.freeswitch.org/20445 >> FS: http://pastebin.freeswitch.org/20444 >> >> Why is Asterisk snappier? >> >> I set all log levels to 0 on my FS box? >> >> Thanks for any ideas. >> >> Best, >> Emrah >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From msc at freeswitch.org Tue Jan 15 04:12:19 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jan 2013 17:12:19 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: Message-ID: What does the FS log say? You should have some mod_spandsp log lines in there that mention what's going on with the fax attempt. Also, are you routing the call to the txfax app? Just curious how the fax call gets connected in the first place. -MC On Mon, Jan 14, 2013 at 4:23 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > Thank you guys, I just sorted out the multiple sofia profiles and have FS > accepting SIP registrations on our LAN. I plugged a phone into the LAN and > it works. > > Now I'm trying to get this VoIP Plug-In for Microsoft Fax to work, that I > downloaded from www.faxback.com. I configured it with a directory number > I created, the dialplan seems to work, but it fails when sending the test > FAX. > > The FAX number I'm calling is a POTS line connected to a Fax modem, which > is answered by Hylafax. This works and is used everyday. The faxback > software is installed on a Windows 7 machine, and configured by putting in > a proxy address of 192.168.1.17, a username and password, and address of > user at domain. > > The test call rings the fax modem, it answers, but then after some > handshaking, it hangs up and the window says the call was hung up. > > Hylafax says: "No answer (T.30 T1 timeout)" > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/b2d6cc01/attachment.html From msc at freeswitch.org Tue Jan 15 04:18:04 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Jan 2013 17:18:04 -0800 Subject: [Freeswitch-users] Is Asterisk snappier than FreeSWITCH to set up calls? In-Reply-To: References: Message-ID: Good to know there's an actual bug open and people are commenting on it. :) The OP (Yehavi B.) seems to have gotten past the issue with a newer commit. That begs the next question: have you gents been able to reproduce this issue on latest git head? Also, was this issue not occurring at some point in the relatively recent past? Reason I ask is that if we can do a git bisect to find the offending commit then that would definitely make troubleshooting a lot easier... Thanks, MC On Mon, Jan 14, 2013 at 5:04 PM, Kristian Kielhofner wrote: > Michael, > > Our experience with that issue is very real. > > I provided details in my comment but call setup time jumps to almost > 2 seconds: > > > http://jira.freeswitch.org/browse/FS-4910?focusedCommentId=38458&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-38458 > > On Mon, Jan 14, 2013 at 7:10 PM, Michael Collins > wrote: > > Assuming that both servers are absolutely identical and running under the > > exact same amount of load, i.e. that "server resources" are not a factor, > > then it looks like Sofia's dialog is slightly chattier and a fraction of > a > > second slower than the equivalent dialog on Asterisk. Keep in mind, > though, > > that this is for a single call. Send 20, 30, 40 CPS or more for an > extended > > period of time and see what the call setup times are. I suspect one of > your > > servers will scale a lot more gracefully than the other... > > > > -MC > > > > On Mon, Jan 14, 2013 at 3:49 PM, Emrah wrote: > >> > >> Ever since I converted to FS, I consistently felt a little delay in > >> setting up my calls. I thought it was just me being too picky or that > the > >> roundtrip required for the digest auth was the cause? > >> Since I now learned that Digest is also required by Asterisk, I am out > of > >> ideas? > >> > >> I tested both SIP servers in parallel and captured a trace. > >> On my Asterisk, the extension 114 dials an IAX peer. The kind of thing > >> that should add more delay? > >> On my FS box, it's milliwatt, executed immediately. > >> > >> Asterisk: http://pastebin.freeswitch.org/20445 > >> FS: http://pastebin.freeswitch.org/20444 > >> > >> Why is Asterisk snappier? > >> > >> I set all log levels to 0 on my FS box? > >> > >> Thanks for any ideas. > >> > >> Best, > >> Emrah > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/9173f17d/attachment-0001.html From curriegrad2004 at gmail.com Tue Jan 15 04:33:27 2013 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 14 Jan 2013 17:33:27 -0800 Subject: [Freeswitch-users] mod_amr transcoing for testing purposes In-Reply-To: References: Message-ID: It's more like they want to make money off of big settlements rather than licensing it out. On Jan 14, 2013 1:13 PM, "Ken Rice" wrote: > Actually its crazy rediculous... Companies like Sangoma cant even license > it. That requirement for integrators isnt sangoma's, its VoiceAge's > requirement > > > > On 1/14/13 2:57 PM, "Kristian Kielhofner" wrote: > > > My understanding is that AMR licensing is even more ridiculous than G729. > > > > On Mon, Jan 14, 2013 at 3:39 PM, Steven Ayre > wrote: > >> Ah, damn you're right Kristian! > >> > >> The data sheet > >> http://sangoma.com/assets/docs/datasheets/en/d100.pdf > >> contains this footnode that is omitted from the web page: > >> > >> "Except for AMR and AMR-WB. Integrators must clear patent license with > >> VoiceAge Corp." > >> > >> :( > >> > >> In which case anyone obtaining licenses from VoiceAge Corp should be > >> free to use them with any implementation... > >> > >> I guess this is due to the same patent pool mess Mike mentioned? > >> > >> -Steve > >> > >> > >> On 14 January 2013 20:34, Steven Ayre wrote: > >>> "All codecs are fully indemnified; no additional licensing is required > >>> for their use." > >>> Source: > >>> > http://sangoma.com/products/media_processing/voice_transcoding_boards/d150.h > >>> tml > >>> > >>> > >>> On 14 January 2013 19:58, Kristian Kielhofner > wrote: > >>>> IIRC Sangoma doesn't provide licensing for AMR and AMR-WB (just the > >>>> implementation). > >>>> > >>>> AFAIK mod_sangoma_codec is no more legal than libamr. > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/ec38ce78/attachment.html From jmesquita at freeswitch.org Tue Jan 15 04:52:01 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 14 Jan 2013 22:52:01 -0300 Subject: [Freeswitch-users] Is Asterisk snappier than FreeSWITCH to set up calls? In-Reply-To: References: Message-ID: MC, apparently, Kristian already took the time to bisect as for his comments on the Jira. If it is fixed on latest head, maybe we don't even need to look at it anymore since it is not even stable branch? Regards, JM Jo?o Mesquita FreeSWITCH? Solutions On Mon, Jan 14, 2013 at 11:18 PM, Michael Collins wrote: > Good to know there's an actual bug open and people are commenting on it. > :) The OP (Yehavi B.) seems to have gotten past the issue with a newer > commit. That begs the next question: have you gents been able to reproduce > this issue on latest git head? Also, was this issue not occurring at some > point in the relatively recent past? Reason I ask is that if we can do a > git bisect to find the offending commit then that would definitely make > troubleshooting a lot easier... > > Thanks, > MC > > > On Mon, Jan 14, 2013 at 5:04 PM, Kristian Kielhofner wrote: > >> Michael, >> >> Our experience with that issue is very real. >> >> I provided details in my comment but call setup time jumps to almost >> 2 seconds: >> >> >> http://jira.freeswitch.org/browse/FS-4910?focusedCommentId=38458&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-38458 >> >> On Mon, Jan 14, 2013 at 7:10 PM, Michael Collins >> wrote: >> > Assuming that both servers are absolutely identical and running under >> the >> > exact same amount of load, i.e. that "server resources" are not a >> factor, >> > then it looks like Sofia's dialog is slightly chattier and a fraction >> of a >> > second slower than the equivalent dialog on Asterisk. Keep in mind, >> though, >> > that this is for a single call. Send 20, 30, 40 CPS or more for an >> extended >> > period of time and see what the call setup times are. I suspect one of >> your >> > servers will scale a lot more gracefully than the other... >> > >> > -MC >> > >> > On Mon, Jan 14, 2013 at 3:49 PM, Emrah wrote: >> >> >> >> Ever since I converted to FS, I consistently felt a little delay in >> >> setting up my calls. I thought it was just me being too picky or that >> the >> >> roundtrip required for the digest auth was the cause? >> >> Since I now learned that Digest is also required by Asterisk, I am out >> of >> >> ideas? >> >> >> >> I tested both SIP servers in parallel and captured a trace. >> >> On my Asterisk, the extension 114 dials an IAX peer. The kind of thing >> >> that should add more delay? >> >> On my FS box, it's milliwatt, executed immediately. >> >> >> >> Asterisk: http://pastebin.freeswitch.org/20445 >> >> FS: http://pastebin.freeswitch.org/20444 >> >> >> >> Why is Asterisk snappier? >> >> >> >> I set all log levels to 0 on my FS box? >> >> >> >> Thanks for any ideas. >> >> >> >> Best, >> >> Emrah >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Michael S Collins >> > Twitter: @mercutioviz >> > http://www.FreeSWITCH.org >> > http://www.ClueCon.com >> > http://www.OSTAG.org >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Kristian Kielhofner >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/5032ce85/attachment-0001.html From krice at freeswitch.org Tue Jan 15 05:04:35 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 14 Jan 2013 20:04:35 -0600 Subject: [Freeswitch-users] Is Asterisk snappier than FreeSWITCH to set up calls? In-Reply-To: Message-ID: No its not fixed... The code pointed to on his bisect is some DB code that hasn?t changed in a while... He?s following up with more info on it shortly... If anyone else is seeing a delay, can you enable presence-debug (in autoload_configs/sofia.xml ) and provide us full debug logs for that? K On 1/14/13 7:52 PM, "Jo?o Mesquita" wrote: > MC, apparently, Kristian already took the time to bisect as for his comments > on the Jira. > > If it is fixed on latest head, maybe we don't even need to look at it anymore > since it is not even stable branch? > > Regards, > JM > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Mon, Jan 14, 2013 at 11:18 PM, Michael Collins wrote: >> Good to know there's an actual bug open and people are commenting on it. :) >> The OP (Yehavi B.) seems to have gotten past the issue with a newer commit. >> That begs the next question: have you gents been able to reproduce this issue >> on latest git head? Also, was this issue not occurring at some point in the >> relatively recent past? Reason I ask is that if we can do a git bisect to >> find the offending commit then that would definitely make troubleshooting a >> lot easier... >> >> Thanks, >> MC >> >> >> On Mon, Jan 14, 2013 at 5:04 PM, Kristian Kielhofner >> wrote: >>> Michael, >>> >>> ? Our experience with that issue is very real. >>> >>> ? I provided details in my comment but call setup time jumps to almost >>> 2 seconds: >>> >>> http://jira.freeswitch.org/browse/FS-4910?focusedCommentId=38458&page=com.at >>> lassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-38458 >>> >>> On Mon, Jan 14, 2013 at 7:10 PM, Michael Collins wrote: >>>> > Assuming that both servers are absolutely identical and running under the >>>> > exact same amount of load, i.e. that "server resources" are not a factor, >>>> > then it looks like Sofia's dialog is slightly chattier and a fraction of a >>>> > second slower than the equivalent dialog on Asterisk. Keep in mind, >>>> though, >>>> > that this is for a single call. Send 20, 30, 40 CPS or more for an >>>> extended >>>> > period of time and see what the call setup times are. I suspect one of >>>> your >>>> > servers will scale a lot more gracefully than the other... >>>> > >>>> > -MC >>>> > >>>> > On Mon, Jan 14, 2013 at 3:49 PM, Emrah wrote: >>>>> >> >>>>> >> Ever since I converted to FS, I consistently felt a little delay in >>>>> >> setting up my calls. I thought it was just me being too picky or that >>>>> the >>>>> >> roundtrip required for the digest auth was the cause? >>>>> >> Since I now learned that Digest is also required by Asterisk, I am out of >>>>> >> ideas? >>>>> >> >>>>> >> I tested both SIP servers in parallel and captured a trace. >>>>> >> On my Asterisk, the extension 114 dials an IAX peer. The kind of thing >>>>> >> that should add more delay? >>>>> >> On my FS box, it's milliwatt, executed immediately. >>>>> >> >>>>> >> Asterisk: http://pastebin.freeswitch.org/20445 >>>>> >> FS: http://pastebin.freeswitch.org/20444 >>>>> >> >>>>> >> Why is Asterisk snappier? >>>>> >> >>>>> >> I set all log levels to 0 on my FS box? >>>>> >> >>>>> >> Thanks for any ideas. >>>>> >> >>>>> >> Best, >>>>> >> Emrah >>>>> >> >>>>> _________________________________________________________________________ >>>>> >> Professional FreeSWITCH Consulting Services: >>>>> >> consulting at freeswitch.org >>>>> >> http://www.freeswitchsolutions.com >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> Official FreeSWITCH Sites >>>>> >> http://www.freeswitch.org >>>>> >> http://wiki.freeswitch.org >>>>> >> http://www.cluecon.com >>>>> >> >>>>> >> FreeSWITCH-users mailing list >>>>> >> FreeSWITCH-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>> > >>>> > >>>> > >>>> > >>>> > -- >>>> > Michael S Collins >>>> > Twitter: @mercutioviz >>>> > http://www.FreeSWITCH.org >>>> > http://www.ClueCon.com >>>> > http://www.OSTAG.org >>>> > >>>> > >>>> > _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>> >>> >>> >>> -- >>> Kristian Kielhofner >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/99aa823c/attachment.html From cmason at frontiernetworks.ca Tue Jan 15 07:19:46 2013 From: cmason at frontiernetworks.ca (Colin Mason) Date: Mon, 14 Jan 2013 23:19:46 -0500 Subject: [Freeswitch-users] MySQL AutoCommit Message-ID: <0D1C698866F66045A6201FD0F59CAC900146FB2EF2@EX.frontier.local> I see this in FreeSWITCH logs whenever MySQL is restarted. Is this something to be concerned about? Using: FreeSWITCH Version 1.3.13b+git~20130110T184435Z~f913670e27 (git f913670 2013-01-10 18:44:35Z) 2013-01-14 06:02:51.007527 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:35.507511 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:44.627524 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:44.767535 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:47.447532 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:50.407523 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:53.527518 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:32.267507 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:41.507535 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:41.547531 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:47.487519 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:50.447521 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:53.567526 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130114/b8a7af58/attachment.html From yehavi.bourvine at gmail.com Tue Jan 15 08:53:46 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 15 Jan 2013 07:53:46 +0200 Subject: [Freeswitch-users] Is Asterisk snappier than FreeSWITCH to set up calls? In-Reply-To: References: Message-ID: As written in FS-4910 my original problem has been resolved. To be sure I've downloaded the latest git (about 30 minutes ago) and tried it - works ok in my environment. Regards, __Yehavi: 2013/1/15 Ken Rice > No its not fixed... The code pointed to on his bisect is some DB code > that hasn?t changed in a while... > > He?s following up with more info on it shortly... > > If anyone else is seeing a delay, can you enable presence-debug (in > autoload_configs/sofia.xml ) and provide us full debug logs for that? > > K > > > > > On 1/14/13 7:52 PM, "Jo?o Mesquita" wrote: > > MC, apparently, Kristian already took the time to bisect as for his > comments on the Jira. > > If it is fixed on latest head, maybe we don't even need to look at it > anymore since it is not even stable branch? > > Regards, > JM > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Mon, Jan 14, 2013 at 11:18 PM, Michael Collins > wrote: > > Good to know there's an actual bug open and people are commenting on it. > :) The OP (Yehavi B.) seems to have gotten past the issue with a newer > commit. That begs the next question: have you gents been able to reproduce > this issue on latest git head? Also, was this issue not occurring at some > point in the relatively recent past? Reason I ask is that if we can do a > git bisect to find the offending commit then that would definitely make > troubleshooting a lot easier... > > Thanks, > MC > > > On Mon, Jan 14, 2013 at 5:04 PM, Kristian Kielhofner > wrote: > > Michael, > > Our experience with that issue is very real. > > I provided details in my comment but call setup time jumps to almost > 2 seconds: > > > http://jira.freeswitch.org/browse/FS-4910?focusedCommentId=38458&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-38458 > > On Mon, Jan 14, 2013 at 7:10 PM, Michael Collins > wrote: > > Assuming that both servers are absolutely identical and running under the > > exact same amount of load, i.e. that "server resources" are not a factor, > > then it looks like Sofia's dialog is slightly chattier and a fraction of > a > > second slower than the equivalent dialog on Asterisk. Keep in mind, > though, > > that this is for a single call. Send 20, 30, 40 CPS or more for an > extended > > period of time and see what the call setup times are. I suspect one of > your > > servers will scale a lot more gracefully than the other... > > > > -MC > > > > On Mon, Jan 14, 2013 at 3:49 PM, Emrah wrote: > >> > >> Ever since I converted to FS, I consistently felt a little delay in > >> setting up my calls. I thought it was just me being too picky or that > the > >> roundtrip required for the digest auth was the cause? > >> Since I now learned that Digest is also required by Asterisk, I am out > of > >> ideas? > >> > >> I tested both SIP servers in parallel and captured a trace. > >> On my Asterisk, the extension 114 dials an IAX peer. The kind of thing > >> that should add more delay? > >> On my FS box, it's milliwatt, executed immediately. > >> > >> Asterisk: http://pastebin.freeswitch.org/20445 > >> FS: http://pastebin.freeswitch.org/20444 > >> > >> Why is Asterisk snappier? > >> > >> I set all log levels to 0 on my FS box? > >> > >> Thanks for any ideas. > >> > >> Best, > >> Emrah > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Ken > > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > * > irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/a4233835/attachment-0001.html From wangd at alongtechnology.com.cn Tue Jan 15 05:25:03 2013 From: wangd at alongtechnology.com.cn (=?gb2312?B?zfW1zw==?=) Date: Tue, 15 Jan 2013 10:25:03 +0800 Subject: [Freeswitch-users] How to pause and resume play an audio file in conference? Message-ID: <201301151025021862606@alongtechnology.com.cn> I will try uuid_fileman on a channel. How to file the wihsklist Jira? Create a improvement issue or new feature issue on Jira? 1. Re: How to pause and resume play an audio file in conference? (Steven Ayre) ------------------------------------------------------------ From: Steven Ayre To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to pause and resume play an audio filein conference? Date: Mon14 Jan 2013 12:51:14 +0000 I suspect it needs a new feature to be added to mod_conference. You could file a wishlist Jira. Or as a workaround, originate a channel that calls into the conference and play the audio on that channel. You'd then be able to use uuid_fileman on that channel. -Steve On 14 January 2013 12:50, Steven Ayre wrote: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/cd5c9def/attachment.html From =?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?= Tue Jan 15 07:31:20 2013 From: =?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?= (=?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?=) Date: Tue, 15 Jan 2013 10:31:20 +0600 Subject: [Freeswitch-users] VOPTech gateways Message-ID: <20130115103120.354a6c1d@mail.btcom.kz> Does anybody know anything about gateways from VOPTech, namely VG1X, VG4X or VG5X? How are they in general, and with FS (inter-operation) in particular? And one more question: there are a couple of companies on the market, which have similar product lines: New Rock Technologies with MX gateways and Real Tone Technologies with their WSS gateways? /Y From =?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?= Tue Jan 15 08:33:40 2013 From: =?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?= (=?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?=) Date: Tue, 15 Jan 2013 11:33:40 +0600 Subject: [Freeswitch-users] Unloop extension Message-ID: <20130115113340.3ccb2287@mail.btcom.kz> What unloop extension is used for? BTW: vars.xml pre-(process)-sets unroll_loops, whereas DPs (public and default) are using channel variable syntax? Is this ok? Y. From hynek.cihlar at gmail.com Tue Jan 15 10:21:53 2013 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Tue, 15 Jan 2013 08:21:53 +0100 Subject: [Freeswitch-users] Early media handling on outbound call In-Reply-To: References: Message-ID: Exactly. Here's the captured SIP flow: https://dl.dropbox.com/u/6276683/sip_flow.png. Between the time 4 and 7, the mobile phone is hung up. After the the time 28 the mobile phone starts ringing again and hung up, the same on times 51 and 74. I am starting to think that this is something my VoIP operator should deal with. What should be the expected flow? Status 480 right after the early media is played back? Hynek On Tue, Jan 15, 2013 at 12:18 AM, Michael Collins wrote: > So you're saying that the provider is sending inband audio in early media > saying that the call is not answering, and then on the same call you hear > ringing? > > -MC > > On Sun, Jan 13, 2013 at 6:28 AM, Hynek Cihlar wrote: > >> Scenario: >> A call is originated to a mobile network. Mobile phone starts ringing and >> is hung up. The mobile phone leg is pre-answered and early media is >> returned saying that the caller is not answering. The message is playbed >> back twice, than silence for cca two seconds and then ringing tone with the >> actual phone starting to ring as well. The phone is hung up, early media >> played back, etc... >> >> The freeswitch console command used to originate the call is as follows: >> originate sofia/internal/1%terminal_domain >> &bridge({origination_caller_id_number=> number>}sofia/gateway/axfone/) >> >> My question is. Is this expected behavior? Should not the destination >> server end the call after early media is played back? Or is it expected >> from me to detect a silence and end the call? If so, what would be the best >> approach to solve it, wait_for_silence? >> >> Thanks for comments! >> Hynek >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/e6d09c73/attachment.html From sertys at gmail.com Tue Jan 15 13:28:26 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Tue, 15 Jan 2013 11:28:26 +0100 Subject: [Freeswitch-users] Chatplan: is msg received success? In-Reply-To: References: Message-ID: Thanks for that tip, alex. Where is the full doc for mod_chat anyway? On Jan 14, 2013 5:19 PM, "Carlos Flor" wrote: > Alex, > > You need to set blocking=true in the chatplan. This will include a > Delivery-Failure=true/false header in the event that is generated when you > send the message. You will need to have an ESL client parse that field to > determine next steps. > > If you don't explicitly set blocking to true, then FS doesn't wait for a > response and so it has no way of knowing if the message was delivered or > not. For that reason, the Delivery-Failure header is not present in the > event when in non-blocking mode. > > > On Mon, Jan 14, 2013 at 6:53 AM, Pip Live wrote: > >> Hi, I use Freeswitch to handle SIP message between SIP extentions, how >> can server know does the message be received by client or not? I noticed >> even if a SIP extention is offline/unregister, SIP message will still be >> sent by server but just the receiver will never get it (even if later the >> receiver online/registered again). >> >> What I want to achieve is: send SIP message to an extention, if received >> OK, nothing happen; if not, forward the message to user's email address. >> >> brgds, Alex >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/d707b4e7/attachment-0001.html From steveayre at gmail.com Tue Jan 15 14:57:52 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 15 Jan 2013 11:57:52 +0000 Subject: [Freeswitch-users] Early media handling on outbound call In-Reply-To: References: Message-ID: It seems odd that they immediately announce that it's not answering then starts ringing. That's sounds like a problem with the VoIP operator - you'll need to report it to them. Is the ringback tone being sent by the operator or generated locally (eg by your softphone) when 180 Ringing is received? It would be possible to set ignore_early_media=true which'd stop you seeing anything until the 480. But in that case you'd also never hear any ringback tone - unless you generated it on FS and in that case it wouldn't necessarily mean the phone was actually ringing. -Steve On 15 January 2013 07:21, Hynek Cihlar wrote: > Exactly. Here's the captured SIP flow: > https://dl.dropbox.com/u/6276683/sip_flow.png. Between the time 4 and 7, the > mobile phone is hung up. After the the time 28 the mobile phone starts > ringing again and hung up, the same on times 51 and 74. > > I am starting to think that this is something my VoIP operator should deal > with. What should be the expected flow? Status 480 right after the early > media is played back? > > Hynek > > > On Tue, Jan 15, 2013 at 12:18 AM, Michael Collins > wrote: >> >> So you're saying that the provider is sending inband audio in early media >> saying that the call is not answering, and then on the same call you hear >> ringing? >> >> -MC >> >> On Sun, Jan 13, 2013 at 6:28 AM, Hynek Cihlar >> wrote: >>> >>> Scenario: >>> A call is originated to a mobile network. Mobile phone starts ringing and >>> is hung up. The mobile phone leg is pre-answered and early media is returned >>> saying that the caller is not answering. The message is playbed back twice, >>> than silence for cca two seconds and then ringing tone with the actual phone >>> starting to ring as well. The phone is hung up, early media played back, >>> etc... >>> >>> The freeswitch console command used to originate the call is as follows: >>> originate sofia/internal/1%terminal_domain >>> &bridge({origination_caller_id_number=>> number>}sofia/gateway/axfone/) >>> >>> My question is. Is this expected behavior? Should not the destination >>> server end the call after early media is played back? Or is it expected from >>> me to detect a silence and end the call? If so, what would be the best >>> approach to solve it, wait_for_silence? >>> >>> Thanks for comments! >>> Hynek >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveayre at gmail.com Tue Jan 15 14:59:38 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 15 Jan 2013 11:59:38 +0000 Subject: [Freeswitch-users] How to pause and resume play an audio file in conference? In-Reply-To: <201301151025021862606@alongtechnology.com.cn> References: <201301151025021862606@alongtechnology.com.cn> Message-ID: I wouldn't worry about that too much - either way it'll get filed and seen by the developers. Since mod_conference already exists and already has a playback feature I'd be inclined to call it an 'improvement'. -Steve On 15 January 2013 02:25, ?? wrote: > I will try uuid_fileman on a channel. > How to file the wihsklist Jira? Create a improvement issue or new feature > issue on Jira? > > > ________________________________ > 1. Re: How to pause and resume play an audio file in conference? > (Steven Ayre) > ------------------------------------------------------------ > From: Steven Ayre > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to pause and resume play an audio > filein conference? > Date: Mon14 Jan 2013 12:51:14 +0000 > I suspect it needs a new feature to be added to mod_conference. You could > file a wishlist Jira. > Or as a workaround, originate a channel that calls into the conference and > play the audio on that channel. You'd then be able to use uuid_fileman on > that channel. > -Steve > On 14 January 2013 12:50, Steven Ayre wrote: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rnbrady at gmail.com Tue Jan 15 16:39:23 2013 From: rnbrady at gmail.com (Richard Brady) Date: Tue, 15 Jan 2013 13:39:23 +0000 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension In-Reply-To: References: <23602598.3790.1358184950672.JavaMail.javamailuser@localhost> <15959415.3795.1358184967040.JavaMail.javamailuser@localhost> Message-ID: I agree with OP. Nothing beats sofia siptrace interleaved with FS logs on the console for debugging FS dialplans, apps, etc. Having the filter would certainly be nifty. I love Homer / Palladion but more for monitoring multiple servers interacting with each other. On 15 January 2013 00:16, Michael Collins wrote: > > > On Mon, Jan 14, 2013 at 10:34 AM, Dmitry Sytchev wrote: > >> We started development of web-ui rather as programming exercise in JS, >> twitter bootstrap and perl Mojolicious :) >> > Perl mojolicious is very cool, btw. > -MC > > >> >> Yes, engine itself does all stuff we need to analyze calls, it stores >> data in db, can be set up to use multiple sensors and store their trace >> data in same DB. Obviously, pcaps and other large raw media data is stored >> locally on sensor machines. >> >> We use it to do full monitoring of all legs of all VoIP calls in our >> network. >> >> >> >> 2013/1/14 Raimund Sacherer >> >>> Thank you Dmitry, >>> >>> are you saying you only use the opensource capture part from >>> voip-monitor on all your 10 FS servers and you do not use their >>> webinterface at all? If I understood this correctly, this opensource does >>> all the capturing and it calculates MOS and Loss scores and stores it in a >>> central MySQL database? >>> >>> That sound's great, I thought the MOS and Loss calculations are in their >>> proprietery part, but if it's included in the opensource part it could be >>> interesting to create your own web-visualization. On the other hand their >>> licence seems to be fairly cheap, some 480 euros ... but I am not sure if >>> this is per server you monitor, or per server you deploy the web-interface >>> to. >>> >>> best, >>> Rai >>> >>> ------------------------------ >>> *From: *"Dmitry Sytchev" >>> >>> *To: *"FreeSWITCH Users Help" >>> *Sent: *Viernes, 11 de Enero 2013 14:35:33 >>> *Subject: *Re: [Freeswitch-users] Bounty for - FS-4268 Sip trace with >>> filter for IP or extension >>> >>> >>> We use VoipMonitor open-source part, with homebrew custom WEB-UI. This >>> is a really cool software, it continuously evolves and captures pretty well >>> even large amounts of VoiP traffic. >>> >>> I highly recommend this software, since it doesn't only captures >>> signaling traffic, but also can capture media/rtcp, automatically convert >>> captured calls to wav files or store raw pcaps for further processing. it >>> also can analyze MOS/Loss scores and many many other different parameters >>> and store them in database per call. >>> >>> We use single instance to store data for investigation of calls with >>> problems on more than 10 servers. >>> >>> >>> >>> >>> 2013/1/11 Raimund Sacherer >>> >>>> Does anyone have any experience with voip monitor? >>>> http://www.voipmonitor.org/ >>>> >>>> We will start evaluating it in our environment as we are switching more >>>> and more to SIP providers, therefore I would love to have better >>>> Monitoring. >>>> >>>> What voipmonitor says is that they are able to filter the calls in >>>> realtime with low overhead an if call quality drops, they can alert you, >>>> and they seem to have a rather nice web interface and they also are used by >>>> others to store the RTP's ... >>>> >>>> What seems to be especially nice is that their capture daemon (as they >>>> state it) adds very little overhead to a system, so depending on call >>>> quantity they say it's neglect-able. >>>> >>>> Would love to hear some comparisions between this and HOMER if some can >>>> provide it. >>>> >>>> Thank you, best >>>> Ray >>>> >>>> ------------------------------ >>>> *From: *"Cal Leeming [Simplicity Media Ltd]" < >>>> cal.leeming at simplicitymedialtd.co.uk> >>>> *To: *"FreeSWITCH Users Help" >>>> *Sent: *Domingo, 6 de Enero 2013 21:01:24 >>>> *Subject: *[Freeswitch-users] Bounty for - FS-4268 Sip trace with >>>> filter for IP or extension >>>> >>>> >>>> Could a core dev speculate on how much the expected bounty would be for >>>> this? >>>> >>>> Thanks >>>> >>>> Cal >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> -- >>>> [image: Logitravel.com] Raimund Sacherer >>>> *Sistemas*Agencia de Viajes Online >>>> www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) >>>> Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca >>>> Tel 902 366 847 | Fax 971 213 495 S?guenos en:[image: Facebook de >>>> Logitravel] [image: Twitter de >>>> Logitravel] [image: Blog de Logitravel] [image: >>>> Logitravel en Youtube] [image: >>>> Logitravel en Foursquare] Descarga >>>> nuestras *aplicaciones para m?vil* [image: Logitravel.com]Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene >>>> informaci?n de car?cter confidencial exclusivamente dirigida a su >>>> destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a >>>> terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de >>>> haber recibido este correo electr?nico por error, se ruega notif?quese >>>> inmediatamente esta circunstancia mediante reenv?o a la direcci?n >>>> electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus >>>> datos forman o formar?n parte de un fichero registrado como CLIENTES con >>>> n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de >>>> Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio >>>> Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. >>>> Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a >>>> estos datos que deber? ejercer mediante escrito a la direcci?n >>>> anteriormente citada. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Best regards, >>> >>> Dmitry Sytchev, >>> IT Engineer >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> -- >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Best regards, >> >> Dmitry Sytchev, >> IT Engineer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/ccd43b8a/attachment-0001.html From slava at tangramltd.com Tue Jan 15 19:02:39 2013 From: slava at tangramltd.com (Viacheslav Dubrovskyi) Date: Tue, 15 Jan 2013 18:02:39 +0200 Subject: [Freeswitch-users] dtmf problem In-Reply-To: <50F38BEE.6090804@synacktics.com> References: <50F38BEE.6090804@synacktics.com> Message-ID: <50F57D9F.4050000@tangramltd.com> 14.01.2013 06:39, Tim Bock ?????: > I have a freeswitch server with a digium tdm wildcard. Compiled, > enabled, and configured freetdm and dahdi...all seems to be working fine > (can make/receive calls). > > My problem is with dtmf. When calling from an external pstn number to > freeswitch, dtmf seems to be detected properly and consistently. But > when calling from a cell phone or via another voip line, dtmf doesn't > seem to be detected, or detected only occasionally. > > On the same system, if I call into the voip number, calls from any of > the three external options (landline, cell, sip) all seem to work fine > with dtmf. > > So the problem seems to be something I've misconfigured with the tdm > card. Can anyone point me in the right direction? Hm. Completely the same problem. Can someone help? Here is the log http://pastebin.freeswitch.org/20447 -- WBR, Viacheslav Dubrovskyi -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3746 bytes Desc: ?????????????????????????????????? ?????????????? S/MIME Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/dc1f6ee8/attachment.bin From jgallartm at gmail.com Tue Jan 15 19:36:41 2013 From: jgallartm at gmail.com (Javier Gallart) Date: Tue, 15 Jan 2013 17:36:41 +0100 Subject: [Freeswitch-users] Can't restablish media Message-ID: Hello I was wondering if someone has experienced this situation. I have a call with bypass_media=true; and this call fails with cause 488. I catch that faliure in a lua script where i set bypass_media=false and change the codec via the variable absolute_codec_string, expecting the Sangoma D500 takes care of the trasncoding . This second call is successful, however when a 183 is received at the b-leg, I see this debug message at the console: 2013-01-15 17:24:27.489568 [ERR] switch_ivr.c:1487 Can't re-establsh media on sofia/internal/xxxxxx at y.y.y.y The 183 message is absorbed by FS, and there is no media transmitted back to a-leg (therefore no ringback heared). However the 200 OK is correctly transmitted, and the calls works apparently works fine. Any idea? This happens at least with Freeswitch versions 1.2.1, 1.2.3 and 1.2.5.3 Thanks in advance Javi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/1fc41604/attachment.html From kris at kriskinc.com Tue Jan 15 19:55:52 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 15 Jan 2013 11:55:52 -0500 Subject: [Freeswitch-users] Is Asterisk snappier than FreeSWITCH to set up calls? In-Reply-To: References: Message-ID: We're in the process of testing Anthony's latest patch. I'll follow-up later today. If this resolves the issue I'd expect this patch to make it into stable. On Mon, Jan 14, 2013 at 8:52 PM, Jo?o Mesquita wrote: > MC, apparently, Kristian already took the time to bisect as for his comments > on the Jira. > > If it is fixed on latest head, maybe we don't even need to look at it > anymore since it is not even stable branch? > > Regards, > JM > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Mon, Jan 14, 2013 at 11:18 PM, Michael Collins > wrote: >> >> Good to know there's an actual bug open and people are commenting on it. >> :) The OP (Yehavi B.) seems to have gotten past the issue with a newer >> commit. That begs the next question: have you gents been able to reproduce >> this issue on latest git head? Also, was this issue not occurring at some >> point in the relatively recent past? Reason I ask is that if we can do a git >> bisect to find the offending commit then that would definitely make >> troubleshooting a lot easier... >> >> Thanks, >> MC >> >> >> On Mon, Jan 14, 2013 at 5:04 PM, Kristian Kielhofner >> wrote: >>> >>> Michael, >>> >>> Our experience with that issue is very real. >>> >>> I provided details in my comment but call setup time jumps to almost >>> 2 seconds: >>> >>> >>> http://jira.freeswitch.org/browse/FS-4910?focusedCommentId=38458&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-38458 >>> >>> On Mon, Jan 14, 2013 at 7:10 PM, Michael Collins >>> wrote: >>> > Assuming that both servers are absolutely identical and running under >>> > the >>> > exact same amount of load, i.e. that "server resources" are not a >>> > factor, >>> > then it looks like Sofia's dialog is slightly chattier and a fraction >>> > of a >>> > second slower than the equivalent dialog on Asterisk. Keep in mind, >>> > though, >>> > that this is for a single call. Send 20, 30, 40 CPS or more for an >>> > extended >>> > period of time and see what the call setup times are. I suspect one of >>> > your >>> > servers will scale a lot more gracefully than the other... >>> > >>> > -MC >>> > >>> > On Mon, Jan 14, 2013 at 3:49 PM, Emrah wrote: >>> >> >>> >> Ever since I converted to FS, I consistently felt a little delay in >>> >> setting up my calls. I thought it was just me being too picky or that >>> >> the >>> >> roundtrip required for the digest auth was the cause? >>> >> Since I now learned that Digest is also required by Asterisk, I am out >>> >> of >>> >> ideas? >>> >> >>> >> I tested both SIP servers in parallel and captured a trace. >>> >> On my Asterisk, the extension 114 dials an IAX peer. The kind of thing >>> >> that should add more delay? >>> >> On my FS box, it's milliwatt, executed immediately. >>> >> >>> >> Asterisk: http://pastebin.freeswitch.org/20445 >>> >> FS: http://pastebin.freeswitch.org/20444 >>> >> >>> >> Why is Asterisk snappier? >>> >> >>> >> I set all log levels to 0 on my FS box? >>> >> >>> >> Thanks for any ideas. >>> >> >>> >> Best, >>> >> Emrah >>> >> >>> >> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> > -- >>> > Michael S Collins >>> > Twitter: @mercutioviz >>> > http://www.FreeSWITCH.org >>> > http://www.ClueCon.com >>> > http://www.OSTAG.org >>> > >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> >>> -- >>> Kristian Kielhofner >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From sdevoy at bizfocused.com Tue Jan 15 20:54:20 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 15 Jan 2013 12:54:20 -0500 Subject: [Freeswitch-users] Redundant Dial Plan in JS In-Reply-To: References: <0b1701cdf206$6a567930$3f036b90$@bizfocused.com> Message-ID: <18af01cdf349$58bb6250$0a3226f0$@bizfocused.com> Thanks Ken. I am left with a few questions though. If I use: What happens if the destination number is just plain old BUSY? Does the dialing user hear the fast-busy signal from "primary" when their route is defective and then hear the ring from secondary? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Sunday, January 13, 2013 11:52 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Redundant Dial Plan in JS See http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Fai lover Or see using the bridge command in conjunction with the continue_on_fail channel variable... This is pretty easy to do... Keep in mind that what good ol' ma bell does is try one route and then if it fails, try to re-route around that... That's all the above is showing you how to do K On 1/13/13 9:22 PM, "Sean Devoy" wrote: Hi, This question stems from the fact that I cannot find a VOIP Service Provider (VSP) that can connect outgoing calls truly reliably. I have customers who expect calls to go through just like good old ma-bell did -100% of the time barring natural disasters. You have to admit those guys set the bar pretty high for us. I was using Voip Innovations with decent success, but my users complain every 2 weeks or so and I send the trace to VI. The standard response is always "We have updated our route tables for that area." I switched to Vitelity where I was assured they have redundant paths and this could never happen! I have only confirmed it happening twice in the last 3 weeks! Of course one of those was during the demo of the new service I had just installed for a new customer. So, I give up. My new approach is "VSPs cannot deliver consistent highly reliable service - work around it!" I know I want a dial plan that will check the bridge status and redial using a different gateway (VSP) if it failed (not busy or no answer - failure). I am pretty sure the best/easy approach is a JavaScript "dialer" script to handle this. I have looked in the wiki for such a sample. If it exists I have missed it. So rather than "re-inventing the wheel" I am reaching out to you all. Does anyone have a dialing script that checks for failure and redials on failure or just continues to voicemail on no answer? I will happily test and tweak it and add it to the wiki if you have something to get me started!!! If not I shall take the plunge and get started! My ideal solution will dial, play a message and redial on failure, maybe 2 or three times or play a failure message and email me on excess errors. I am leaning toward "Call congestion, rerouting. Please stand by" while the new bridge is being attempted. Thanks, Sean _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/03540004/attachment-0001.html From schoch+freeswitch.org at xwin32.com Tue Jan 15 20:58:28 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 15 Jan 2013 09:58:28 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: Message-ID: On Mon, Jan 14, 2013 at 5:12 PM, Michael Collins wrote: > What does the FS log say? You should have some mod_spandsp log lines in > there that mention what's going on with the fax attempt. Also, are you > routing the call to the txfax app? Just curious how the fax call gets > connected in the first place. The call is being routed *from* the Microsoft FAX plug in. The routing works like this: 1. FAX is initiated with Microsoft FAX on Windows 2. It goes through the faxback.com plug-in on the same machine. 3. The plug-in is configured to make a SIP connection to the FS pbx. 4. The FS pbx routes the call to the PSTN through sip.flowroute.com. 5. The call is received by a POTS line, connected to a modem. 6. The modem is connected with a serial line to a Linux machine. 7. The Linux machine runs Hylafax (www.hylafax.org). Steps 4-5 work for voice calls, using Polycom phones. Steps 5-7 work for other FAX calls, received often. The log for the faxback.com plug-in is at http://pastebin.freeswitch.org/20448. The FS log is at http://pastebin.freeswitch.org/20449. The Hylafax log is at http://pastebin.freeswitch.org/20450. New info: I put a test set on the PSTN fax line. (Yes, I'm old school telephony.) I hear the fax modem giving the normal terminating fax pre-message signal, and the originating fax sending the normal CNG tones. However, these tones don't stop. It's like the originating fax can't hear the pre-message signal from the fax modem. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/cd1234b4/attachment.html From drk at drkngs.net Tue Jan 15 21:48:55 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 15 Jan 2013 10:48:55 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: Message-ID: <20130115184855.ed09f548@mail.tritonwest.net> It seems that you may be debugging way too many things at once. Do you know much about the FAXBACK.COM internals? Just for a test you know you can bypass it. FreeSWITCH will work just fine on a windows box, using the FAXModem stuff, and com0com driver. That's what I use to have the M$ fax service talk to, and it works great. No hardware, and FS doing T38 right out of the Fax Service Server. At least if you test this way, you have FS logs and access to all the internals, and you can eliminate the "Black Box" in the middle. --Dave _____ From: Steven Schoch [mailto:schoch+freeswitch.org at xwin32.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 15 Jan 2013 09:58:28 -0800 Subject: Re: [Freeswitch-users] FAX problems On Mon, Jan 14, 2013 at 5:12 PM, Michael Collins wrote: What does the FS log say? You should have some mod_spandsp log lines in there that mention what's going on with the fax attempt. Also, are you routing the call to the txfax app? Just curious how the fax call gets connected in the first place. The call is being routed from the Microsoft FAX plug in. The routing works like this: 1. FAX is initiated with Microsoft FAX on Windows 2. It goes through the faxback.com plug-in on the same machine. 3. The plug-in is configured to make a SIP connection to the FS pbx. 4. The FS pbx routes the call to the PSTN through sip.flowroute.com. 5. The call is received by a POTS line, connected to a modem. 6. The modem is connected with a serial line to a Linux machine. 7. The Linux machine runs Hylafax (www.hylafax.org). Steps 4-5 work for voice calls, using Polycom phones. Steps 5-7 work for other FAX calls, received often. The log for the faxback.com plug-in is at http://pastebin.freeswitch.org/20448. The FS log is at http://pastebin.freeswitch.org/20449. The Hylafax log is at http://pastebin.freeswitch.org/20450. New info: I put a test set on the PSTN fax line. (Yes, I'm old school telephony.) I hear the fax modem giving the normal terminating fax pre-message signal, and the originating fax sending the normal CNG tones. However, these tones don't stop. It's like the originating fax can't hear the pre-message signal from the fax modem. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/e8381ac6/attachment.html From hexade at hotmail.com Tue Jan 15 22:05:19 2013 From: hexade at hotmail.com (Adelia C.) Date: Tue, 15 Jan 2013 14:05:19 -0500 Subject: [Freeswitch-users] Temporarily stop/start Freeswitch 200OKs on OPTIONS Ping? Message-ID: Is there a way to do this? I found some old Q&As on this forum that reject this as non-spec compliant (I agree) BUT ... I need to disable 200OKs short period of time. This is my use case: I want to stop new traffic to specific Freeswitch servers (drain them) previous to a new deploy. There are reasons for deploying this way (as opposed to just dropping the new code in) which I will not explain here. For now, I need to do it this way. I don't control the SBC so the easiest way is to stop replying to OPTIONS on the FS servers to go out of commission. The SBC will stop forwarding new calls while handling those in progress. Can I do this? Thank you, Adelia -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/568a0eb1/attachment.html From gozdal at gmail.com Tue Jan 15 22:15:47 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Tue, 15 Jan 2013 20:15:47 +0100 Subject: [Freeswitch-users] Temporarily stop/start Freeswitch 200OKs on OPTIONS Ping? In-Reply-To: References: Message-ID: 2013/1/15 Adelia C. > Is there a way to do this? I found some old Q&As on this forum that reject this as non-spec compliant (I agree) BUT ... I need to disable 200OKs short period of time. > > This is my use case: > > I want to stop new traffic to specific Freeswitch servers (drain them) previous to a new deploy. There are reasons for deploying this way (as opposed to just dropping the new code in) which I will not explain here. For now, I need to do it this way. > > I don't control the SBC so the easiest way is to stop replying to OPTIONS on the FS servers to go out of commission. The SBC will stop forwarding new calls while handling those in progress. I believe you are looking for sip-options-respond-503-on-busy: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-options-respond-503-on-busy: "Setting this param to true is especially useful if you're using a proxy such as OpenSIPS or Kamailio with dispatcher module to probe your FreeSWITCH servers by sending SIP OPTIONS. " Best regards From kris at kriskinc.com Tue Jan 15 22:31:53 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 15 Jan 2013 14:31:53 -0500 Subject: [Freeswitch-users] Temporarily stop/start Freeswitch 200OKs on OPTIONS Ping? In-Reply-To: References: Message-ID: Then you will need to issue "fsctl pause" in case that's not clear from the explanation at this link. On Tue, Jan 15, 2013 at 2:15 PM, Marcin Gozdalik wrote: > 2013/1/15 Adelia C. > >> Is there a way to do this? I found some old Q&As on this forum that reject this as non-spec compliant (I agree) BUT ... I need to disable 200OKs short period of time. >> >> This is my use case: >> >> I want to stop new traffic to specific Freeswitch servers (drain them) previous to a new deploy. There are reasons for deploying this way (as opposed to just dropping the new code in) which I will not explain here. For now, I need to do it this way. >> >> I don't control the SBC so the easiest way is to stop replying to OPTIONS on the FS servers to go out of commission. The SBC will stop forwarding new calls while handling those in progress. > > > I believe you are looking for sip-options-respond-503-on-busy: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-options-respond-503-on-busy: > > "Setting this param to true is especially useful if you're using a > proxy such as OpenSIPS or Kamailio with dispatcher module to probe > your FreeSWITCH servers by sending SIP OPTIONS. " > > Best regards > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From hexade at hotmail.com Tue Jan 15 22:51:37 2013 From: hexade at hotmail.com (Adelia C.) Date: Tue, 15 Jan 2013 14:51:37 -0500 Subject: [Freeswitch-users] Temporarily stop/start Freeswitch 200OKs on OPTIONS Ping? In-Reply-To: References: , , Message-ID: Thank you both! This is exactly what I was looking for! Adelia > Date: Tue, 15 Jan 2013 14:31:53 -0500 > From: kris at kriskinc.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Temporarily stop/start Freeswitch 200OKs on OPTIONS Ping? > > Then you will need to issue "fsctl pause" in case that's not clear > from the explanation at this link. > > On Tue, Jan 15, 2013 at 2:15 PM, Marcin Gozdalik wrote: > > 2013/1/15 Adelia C. > > > >> Is there a way to do this? I found some old Q&As on this forum that reject this as non-spec compliant (I agree) BUT ... I need to disable 200OKs short period of time. > >> > >> This is my use case: > >> > >> I want to stop new traffic to specific Freeswitch servers (drain them) previous to a new deploy. There are reasons for deploying this way (as opposed to just dropping the new code in) which I will not explain here. For now, I need to do it this way. > >> > >> I don't control the SBC so the easiest way is to stop replying to OPTIONS on the FS servers to go out of commission. The SBC will stop forwarding new calls while handling those in progress. > > > > > > I believe you are looking for sip-options-respond-503-on-busy: > > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#sip-options-respond-503-on-busy: > > > > "Setting this param to true is especially useful if you're using a > > proxy such as OpenSIPS or Kamailio with dispatcher module to probe > > your FreeSWITCH servers by sending SIP OPTIONS. " > > > > Best regards > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/99fc0e3d/attachment-0001.html From jtbock at synacktics.com Tue Jan 15 22:39:53 2013 From: jtbock at synacktics.com (Tim Bock) Date: Tue, 15 Jan 2013 12:39:53 -0700 Subject: [Freeswitch-users] dtmf problem In-Reply-To: <50F57D9F.4050000@tangramltd.com> References: <50F38BEE.6090804@synacktics.com> <50F57D9F.4050000@tangramltd.com> Message-ID: <50F5B089.3060000@synacktics.com> I can tell you how I solved the problem I was having. Don't know if this is preferred, or if there is a better way, but I'm now able to detect dtmf from all sources (pstn, cell, sip). Apparently, mod spandsp has better dtmf detection than the dtmf detection in the freeswitch core. On the wiki page for spandsp (http://wiki.freeswitch.org/wiki/Mod_spandsp), there is a section on dtmf. I added the spandsp_start_dtmf parameter before calling the ivr, and that fixed my issue. Hope this helps, Tim On 01/15/2013 09:02 AM, Viacheslav Dubrovskyi wrote: > 14.01.2013 06:39, Tim Bock ?????: >> I have a freeswitch server with a digium tdm wildcard. Compiled, >> enabled, and configured freetdm and dahdi...all seems to be working fine >> (can make/receive calls). >> >> My problem is with dtmf. When calling from an external pstn number to >> freeswitch, dtmf seems to be detected properly and consistently. But >> when calling from a cell phone or via another voip line, dtmf doesn't >> seem to be detected, or detected only occasionally. >> >> On the same system, if I call into the voip number, calls from any of >> the three external options (landline, cell, sip) all seem to work fine >> with dtmf. >> >> So the problem seems to be something I've misconfigured with the tdm >> card. Can anyone point me in the right direction? > Hm. Completely the same problem. > > Can someone help? > > Here is the log http://pastebin.freeswitch.org/20447 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Tim Bock Synacktics, LLC www.synacktics.com (505)795-1511 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/a297f7c8/attachment.html From gavin.henry at gmail.com Wed Jan 16 00:03:14 2013 From: gavin.henry at gmail.com (Gavin Henry) Date: Tue, 15 Jan 2013 21:03:14 +0000 Subject: [Freeswitch-users] Status of mod_lua and dbh.release() Message-ID: Hi all, Does this release the db handle like it says? dbh:release() (optional) releases the handle back to the pool so it can be re-used by another thread. This is also automatically done when the dbh goes out of scope and is garbage collected (for example when your script returns). We're seeing quite a free PostgreSQL 9.1 processes left. -- http://www.suretecsystems.com/services/openldap/ http://www.surevoip.co.uk From msc at freeswitch.org Wed Jan 16 02:17:59 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Jan 2013 15:17:59 -0800 Subject: [Freeswitch-users] Is Asterisk snappier than FreeSWITCH to set up calls? In-Reply-To: References: Message-ID: On Mon, Jan 14, 2013 at 5:52 PM, Jo?o Mesquita wrote: > MC, apparently, Kristian already took the time to bisect as for his > comments on the Jira. > > If it is fixed on latest head, maybe we don't even need to look at it > anymore since it is not even stable branch? > This is the exact reason we need bug marshals. Someone needs to dig and answer this valid question. We will be going over a few of these on tomorrow's conference call. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/ca9fed1a/attachment.html From itsusama at gmail.com Wed Jan 16 03:24:51 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Wed, 16 Jan 2013 05:24:51 +0500 Subject: [Freeswitch-users] Mod_Conference Improvement (Denoise) Message-ID: Hi, I talked about this a while back on the channel, the way how energy-level is implemented in mod_conference isn't the right way to handle noise in my opinion because the participants that get bridged would have a different noise floor depending on where they are located, so if I set the value of energy-level too high to accommodate one participant, any other participant in a quite sitting would never get bridged, I proposed we should tackle this the way Asterisk does using a denoise dialplan function ( https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE), we can use libav for noise filtering on the muxed audio. I'm willing to contribute if one of the devs would help me out regarding implementation. Any comments? -Regards -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/1e762047/attachment.html From schoch+freeswitch.org at xwin32.com Wed Jan 16 03:39:15 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 15 Jan 2013 16:39:15 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: <20130115184855.ed09f548@mail.tritonwest.net> References: <20130115184855.ed09f548@mail.tritonwest.net> Message-ID: My whole plan here is to replace the phone system currently in the office (a NexPath system, a really nice PC-chassis pbx system from the 1990's) to a more modern VoIP system, and git rid of all our POTS lines. But I need a simple solution for the users who send FAXes from their Windows systems. On Tue, Jan 15, 2013 at 10:48 AM, Dave R. Kompel wrote: > ** > It seems that you may be debugging way too many things at once. Do you > know much about the FAXBACK.COM internals? > Yes, that's probably correct. No, I don't know much about faxback.com. However, I just tried a different one, www.t38faxvoip.com. It's a much more involved Windows installation, but it came up with the same symptoms. This is leading me to believe that the problem is more likely my FS configuration. Just for a test you know you can bypass it. FreeSWITCH will work just fine > on a windows box, using the FAXModem stuff, and com0com driver. > That's not my objective. I'm still early in the learning process for FreeSwitch, and I'd like to keep it to one installation, on an OS that I can understand (Linux). That's what I use to have the M$ fax service talk to, and it works great. > No hardware, and FS doing T38 right out of the Fax Service Server. > Let me see if I understand: MS Fax is talking through a virtual serial cable (com0com) to what it thinks is a FAX modem. On the other side of that virtual serial cable is something that takes the serial FAX data, translates it to T38, and injects it into FS. I'm a little unclear on the last 2 pieces, because it still sounds like a black box to me. What's "Fax Service Server"? On Mon, Jan 14, 2013 at 5:12 PM, Michael Collins wrote: > Also, are you routing the call to the txfax app? I'm not. I assumed that the purpose of txfax is to encode a TIFF file into a T.38 stream, and my source is already T.38. Am I missing something? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/820a775a/attachment.html From msc at freeswitch.org Wed Jan 16 03:54:12 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Jan 2013 16:54:12 -0800 Subject: [Freeswitch-users] Unloop extension In-Reply-To: <20130115113340.3ccb2287@mail.btcom.kz> References: <20130115113340.3ccb2287@mail.btcom.kz> Message-ID: Yes, it's okay. The pre-process does a set and ${unroll_loops} gets set as needed. Then this dp entry does the work: More info on sip_looped_call var: http://wiki.freeswitch.org/wiki/Variable_sip_looped_call -MC On Mon, Jan 14, 2013 at 9:33 PM, ????? ??????? wrote: > What unloop extension is used for? BTW: vars.xml pre-(process)-sets > unroll_loops, whereas DPs (public and default) are using channel variable > syntax? Is this ok? > > Y. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/7f51cecf/attachment-0001.html From msc at freeswitch.org Wed Jan 16 04:03:20 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Jan 2013 17:03:20 -0800 Subject: [Freeswitch-users] Mod_Conference Improvement (Denoise) In-Reply-To: References: Message-ID: While I don't have a problem with the concept of noise filtering I have to point out that each member of the conference can have his or her own noise level setting. From fs_cli: conference energy [] You only need to apply it to the person who is in the noisy location. In fact, in the example configuration the user can dial 9 to increase the energy threshold (for when he's in a noisy environment) or press 7 to decrease the threshold. Pressing 8 will reset the energy to normal, which I believe is 200. -MC On Tue, Jan 15, 2013 at 4:24 PM, Usama Zaidi wrote: > Hi, > > I talked about this a while back on the channel, the way how energy-level > is implemented in mod_conference isn't the right way to handle noise in my > opinion because the participants that get bridged would have a different > noise floor depending on where they are located, so if I set the value of > energy-level too high to accommodate one participant, any other participant > in a quite sitting would never get bridged, I proposed we should tackle > this the way Asterisk does using a denoise dialplan function ( > https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE), > we can use libav for noise filtering on the muxed audio. I'm willing to > contribute if one of the devs would help me out regarding implementation. > Any comments? > > -Regards > > -- > I'd love to change the world, but they wont gimme the source code to it > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/66abf73d/attachment.html From drk at drkngs.net Wed Jan 16 05:52:42 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 15 Jan 2013 18:52:42 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: Message-ID: <20130116025242.83fde4e4@mail.tritonwest.net> In FreeSWITCH, mod_spandsp, will also emulate fax modems. Com0Com is like a PTY driver for windows. When you set modem_count in spandsp.conf.xml to a count other then 0, the windows implmentation starts at COM3. So to get T.38 emulated fax modems, you don't need anything other then FreeSWITCH. Another option would be on the Linux box, enable SPANDSP modems, and use com0com on windows to backup com ports to raw TCP connections. It supports both. That way it would talk over a TCP pipe to an emulated modem on your other FreeSWITCH box. --Dave _____ From: Steven Schoch [mailto:schoch+freeswitch.org at xwin32.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 15 Jan 2013 16:39:15 -0800 Subject: Re: [Freeswitch-users] FAX problems My whole plan here is to replace the phone system currently in the office (a NexPath system, a really nice PC-chassis pbx system from the 1990's) to a more modern VoIP system, and git rid of all our POTS lines. But I need a simple solution for the users who send FAXes from their Windows systems. On Tue, Jan 15, 2013 at 10:48 AM, Dave R. Kompel wrote: It seems that you may be debugging way too many things at once. Do you know much about the FAXBACK.COM internals? Yes, that's probably correct. No, I don't know much about faxback.com. However, I just tried a different one, www.t38faxvoip.com. It's a much more involved Windows installation, but it came up with the same symptoms. This is leading me to believe that the problem is more likely my FS configuration. Just for a test you know you can bypass it. FreeSWITCH will work just fine on a windows box, using the FAXModem stuff, and com0com driver. That's not my objective. I'm still early in the learning process for FreeSwitch, and I'd like to keep it to one installation, on an OS that I can understand (Linux). That's what I use to have the M$ fax service talk to, and it works great. No hardware, and FS doing T38 right out of the Fax Service Server. Let me see if I understand: MS Fax is talking through a virtual serial cable (com0com) to what it thinks is a FAX modem. On the other side of that virtual serial cable is something that takes the serial FAX data, translates it to T38, and injects it into FS. I'm a little unclear on the last 2 pieces, because it still sounds like a black box to me. What's "Fax Service Server"? On Mon, Jan 14, 2013 at 5:12 PM, Michael Collins wrote: Also, are you routing the call to the txfax app? I'm not. I assumed that the purpose of txfax is to encode a TIFF file into a T.38 stream, and my source is already T.38. Am I missing something? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/fe34d073/attachment.html From mike at jerris.com Wed Jan 16 05:58:47 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Jan 2013 21:58:47 -0500 Subject: [Freeswitch-users] Mod_Conference Improvement (Denoise) In-Reply-To: References: Message-ID: <5274F7F1-8BC9-4721-B98F-A163495CD730@jerris.com> If you wanted to implement something like this you could do so as a media bug, it does not need to touch mod_conference in any way. Take a look at mod_ladspa for an example of a similar module that modifies audio using a media bug. Give it a try and see if you can get something working. Mike On Jan 15, 2013, at 8:03 PM, Michael Collins wrote: > While I don't have a problem with the concept of noise filtering I have to point out that each member of the conference can have his or her own noise level setting. From fs_cli: > > conference energy [] > > You only need to apply it to the person who is in the noisy location. In fact, in the example configuration the user can dial 9 to increase the energy threshold (for when he's in a noisy environment) or press 7 to decrease the threshold. Pressing 8 will reset the energy to normal, which I believe is 200. > > -MC > > On Tue, Jan 15, 2013 at 4:24 PM, Usama Zaidi wrote: > Hi, > > I talked about this a while back on the channel, the way how energy-level is implemented in mod_conference isn't the right way to handle noise in my opinion because the participants that get bridged would have a different noise floor depending on where they are located, so if I set the value of energy-level too high to accommodate one participant, any other participant in a quite sitting would never get bridged, I proposed we should tackle this the way Asterisk does using a denoise dialplan function (https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE), we can use libav for noise filtering on the muxed audio. I'm willing to contribute if one of the devs would help me out regarding implementation. Any comments? > > -Regards > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130115/36477f0d/attachment.html From slava at tangramltd.com Wed Jan 16 10:24:44 2013 From: slava at tangramltd.com (Viacheslav Dubrovskyi) Date: Wed, 16 Jan 2013 09:24:44 +0200 Subject: [Freeswitch-users] dtmf problem In-Reply-To: <50F5B089.3060000@synacktics.com> References: <50F38BEE.6090804@synacktics.com> <50F57D9F.4050000@tangramltd.com> <50F5B089.3060000@synacktics.com> Message-ID: <50F655BC.1020608@tangramltd.com> 15.01.2013 21:39, Tim Bock ?????: > I can tell you how I solved the problem I was having. Don't know if > this is preferred, or if there is a better way, but I'm now able to > detect dtmf from all sources (pstn, cell, sip). > > Apparently, mod spandsp has better dtmf detection than the dtmf > detection in the freeswitch core. On the wiki page for spandsp > (http://wiki.freeswitch.org/wiki/Mod_spandsp), there is a section on > dtmf. I added the spandsp_start_dtmf parameter before calling the > ivr, and that fixed my issue. > > Hope this helps, Thank you so much! -- WBR, Viacheslav Dubrovskyi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/ab674c41/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3746 bytes Desc: ?????????????????????????????????? ?????????????? S/MIME Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/ab674c41/attachment-0001.bin From support at ecn.net.au Wed Jan 16 07:02:26 2013 From: support at ecn.net.au (support at ecn.net.au) Date: Wed, 16 Jan 2013 14:02:26 +1000 Subject: [Freeswitch-users] SBC In-band DTMF Message-ID: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> Hi All We're quite new to Freeswitch and are in the process of migrating from OpenSer (as an SBC) to Freeswitch. Mostly all is working well, except an oddity on DTMF. Our scenario: Telco/SIP Provider A is passing us calls using DTMF inband. We have a freeswitch configured as a SBC using 2 sip profiles (telco and internal) to topology hide and manage distribution of calls to the PBX servers located behind the SBC. The freeswitch will be handling up to a few hundred calls so we're trying to keep it lightweight. Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling customer needs. An example inbound call profile looks like this: > Initially when calling into the platform IVR type applications runinng on our PBX boxes would not work (you could hear the DTMF but the platform did not recognise the tones). We have had to add the appliation start_dtmf in order for Freeswitch to pass the DTMF to the Asterisk PBX behind the SBC. Interestingly on our OpenSer platform we just proxied the media (rtpproxy) with inband DTMF from the Telco and our PBX boxes recognised the inband DTMF tones on the PBX platforms and IVR type applications just worked. However under freeswitch if we don't start_dtmf before the bridge the backend PBX boxes don't recognise the DTMF inband (even though the tones are audible ie you can hear them on a call recording on the PBX). Have we missed something here? We would have thought with inband DTMF on non compressed codec (no transcoding) that the tones would just work with the media stream? We have confirmed both legs are PCMA and when using start_dtmf the first second of the call is clipped. Kind Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/835ccd17/attachment.html From =?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?= Wed Jan 16 07:09:14 2013 From: =?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?= (=?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?=) Date: Wed, 16 Jan 2013 10:09:14 +0600 Subject: [Freeswitch-users] Unloop extension Message-ID: <20130116100914.d57f06c7@mail.btcom.kz> Ain't what was set with Yes, it's okay. The pre-process does a set and ${unroll_loops} gets set as > needed. Then this dp entry does the work: > > > > > > > > More info on sip_looped_call var: > http://wiki.freeswitch.org/wiki/Variable_sip_looped_call > > -MC > > On Mon, Jan 14, 2013 at 9:33 PM, ????? ??????? > wrote: > > > What unloop extension is used for? BTW: vars.xml pre-(process)-sets > > unroll_loops, whereas DPs (public and default) are using channel variable > > syntax? Is this ok? > > > > Y. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > From oej at edvina.net Wed Jan 16 11:09:07 2013 From: oej at edvina.net (Olle E. Johansson) Date: Wed, 16 Jan 2013 09:09:07 +0100 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> Message-ID: <7FA4A571-7A65-4493-84A4-90052B9CADC8@edvina.net> 16 jan 2013 kl. 05:02 skrev "support at ecn.net.au" : > We're quite new to Freeswitch and are in the process of migrating from OpenSer (as an SBC) to Freeswitch. > It surprises me that you compare two very different products and migrate from one to the other. OpenSER (or Kamailio.org which is the new name) and FreeSwitch work very well together. FreeSwitch is a b2bua that handles the media and Kamailio kan handle the SIP side and add a layer of protection with modules like Pike that give DoS attack prevention, the SIP pipes for QoS and much more. Don't migrate, use both together! /O Kamailio marketing dept :-) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/d45e761e/attachment.html From haloha201 at gmail.com Wed Jan 16 11:49:34 2013 From: haloha201 at gmail.com (haloha) Date: Wed, 16 Jan 2013 15:49:34 +0700 Subject: [Freeswitch-users] need help on call recording Message-ID: hi all i try to record all calls to/from local extension so my dialplan FS only records the caller voicem FS doesnt record the called voice How do i record the entire calls(caller and called) Thank you Ha` From jaybinks at gmail.com Wed Jan 16 12:17:27 2013 From: jaybinks at gmail.com (jay binks) Date: Wed, 16 Jan 2013 19:17:27 +1000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> Message-ID: on your internal sip profile ( one facing PBX's ) set then note the dialplan context that super dodgey sip carrier ( the one doing inband only ) sends calls to, and use on calls from them. Id advise against in that location unless you REALLY want that. that should do what you want. On 16 January 2013 14:02, support at ecn.net.au wrote: > Hi All > > > > We're quite new to Freeswitch and are in the process of migrating from > OpenSer (as an SBC) to Freeswitch. > > > > Mostly all is working well, except an oddity on DTMF. > > > > Our scenario: > > > > Telco/SIP Provider A is passing us calls using DTMF inband. > > > > We have a freeswitch configured as a SBC using 2 sip profiles (telco and > internal) to topology hide and manage > > distribution of calls to the PBX servers located behind the SBC. > > > > The freeswitch will be handling up to a few hundred calls so we're trying > to keep it lightweight. > > > > Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling > customer needs. > > > > An example inbound call profile looks like this: > > > > > > > > > > > > > > > > > > > > > > Initially when calling into the platform IVR type applications runinng on > our PBX boxes would not > > work (you could hear the DTMF but the platform did not recognise the > tones). > > > > We have had to add the appliation start_dtmf in order for Freeswitch to > pass the DTMF to the Asterisk > > PBX behind the SBC. Interestingly on our OpenSer platform we just > proxied the media (rtpproxy) with > > inband DTMF from the Telco and our PBX boxes recognised the inband DTMF > tones on the PBX platforms and > > IVR type applications just worked. > > > > However under freeswitch if we don't start_dtmf before the bridge the > backend PBX boxes don't recognise > > the DTMF inband (even though the tones are audible ie you can hear them on > a call recording on the > > PBX). > > > > Have we missed something here? We would have thought with inband DTMF on > non compressed codec (no > > transcoding) that the tones would just work with the media stream? > > > > We have confirmed both legs are PCMA and when using start_dtmf the first > second of the call is clipped. > > > > > > > Kind Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/131392b3/attachment-0001.html From rentmycoder at gmail.com Wed Jan 16 12:48:48 2013 From: rentmycoder at gmail.com (rentmycoder rentmycoder) Date: Wed, 16 Jan 2013 10:48:48 +0100 Subject: [Freeswitch-users] limit memory usage of freeswitch Message-ID: Hi All, According to my experineces and some info found on FS lists it is the normal behaviour that freeswitch eats up all of the available memory... It's not a memory leak, but the standard behaviour of FS by design... Correct me if I'm wrong... Is there any way to limit FS memory usage??? I need another services to run on the FS box... Thanks, John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/b0c798c1/attachment.html From jaybinks at gmail.com Wed Jan 16 13:09:04 2013 From: jaybinks at gmail.com (jay binks) Date: Wed, 16 Jan 2013 20:09:04 +1000 Subject: [Freeswitch-users] limit memory usage of freeswitch In-Reply-To: References: Message-ID: OpenVZ / Proxmox will allow you use set quota's and containerize your freeswitch... thats ONE way to do what your after. On 16 January 2013 19:48, rentmycoder rentmycoder wrote: > Hi All, > > According to my experineces and some info found on FS lists it is the > normal behaviour that freeswitch eats up all of the available memory... > It's not a memory leak, but the standard behaviour of FS by design... > Correct me if I'm wrong... > > Is there any way to limit FS memory usage??? > > I need another services to run on the FS box... > > Thanks, > John > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/73c73819/attachment.html From ben122uk at gmail.com Wed Jan 16 13:36:08 2013 From: ben122uk at gmail.com (Ben) Date: Wed, 16 Jan 2013 10:36:08 -0000 Subject: [Freeswitch-users] ZRTP error on call setup when dialling client uses ZRTP Message-ID: <022601cdf3d5$4c8e9a10$e5abce30$@gmail.com> Hi all, My server is running the latest git stable build of Freeswitch, all config is default and server/clients are on simple LAN setup. I'm having an issue when using ZRTP on one client only, specifically the dialling client. The calling device has ZRTP enabled, and the called device does not. In this scenario, the FS server should act as a man in the middle and exchange keys with the calling device. This results in the A leg being encrypted and the B leg un-encrypted. However as soon as the key exchange initiates, the A client errors and tears down the call. What's strange is that when I initiate the call from the non-zrtp client to the zrtp client, the keys exchange fine between server and client A, and 2-way audio is present. So it's only when it's the dialling handset do I have the problem. I've then gone on and tested the same thing on the latest git, and the problem goes away. So my question is, can anyone advise on what changes/patches have been applied to the latest git that has resolved this issue? I've tried to search the git tree, but I'll be honest I'm a bit unfamiliar with how it works and I'm not sure that I'm searching properly. The ZRTP device is an iPhone running Groundwire, and it doesn't seem to matter what codec I use. Thanks for the help! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/b654d3bf/attachment.html From shahzad.bhatti at g-r-v.com Wed Jan 16 14:39:22 2013 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Wed, 16 Jan 2013 16:39:22 +0500 Subject: [Freeswitch-users] configuring Acoustic Model for german language problem Message-ID: Hello Everybody, i am trying to configure Acoustic Model for german language on localhost as wiki link http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Acoustic_Model_for_german_language but when i use ./etc/voxforge_de_sphinx_train.transcription i got error ./etc/voxforge_de_sphinx_train.transcription: line 1: syntax error near unexpected token `(' ./etc/voxforge_de_sphinx_train.transcription: line 1: ` MAN UNTERSCHEIDET ABSCHLUSSFREIHEIT (anonymous-20080117-ihl_de5-040)' any positive reply will be highly appreciated. regards shahzad bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/81e05a40/attachment.html From muqtheear.s at gmail.com Wed Jan 16 15:08:33 2013 From: muqtheear.s at gmail.com (Muqtheear S) Date: Wed, 16 Jan 2013 17:38:33 +0530 Subject: [Freeswitch-users] Issue - playing mp4 video files Message-ID: 1. Please tell me how to convert mpeg4-generic encoded audio data to PCMU encoded audio data in .mp4 file. I tried with ffmpeg, but couldn't succeed. 2. If you have any mp4 file with PCMU sound track, please share with me. 3. Does freeswitch comes with mpeg4-generic codec support? Thanks & Regards, Muqtheear.S -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/b7d07d0a/attachment.html From royj at yandex.ru Wed Jan 16 15:11:08 2013 From: royj at yandex.ru (royj) Date: Wed, 16 Jan 2013 16:11:08 +0400 Subject: [Freeswitch-users] ZRTP error on call setup when dialling client uses ZRTP In-Reply-To: <022601cdf3d5$4c8e9a10$e5abce30$@gmail.com> References: <022601cdf3d5$4c8e9a10$e5abce30$@gmail.com> Message-ID: <20130116161108.9cc0a0da5fb17db3dfd23023@yandex.ru> What is the proxy-media mode on both sides, in both cases. It matters. In my case I have experienced this On Wed, 16 Jan 2013 10:36:08 -0000 "Ben" wrote: > Hi all, > > > > My server is running the latest git stable build of Freeswitch, all config > is default and server/clients are on simple LAN setup. I'm having an issue > when using ZRTP on one client only, specifically the dialling client. > > > > The calling device has ZRTP enabled, and the called device does not. In > this scenario, the FS server should act as a man in the middle and exchange > keys with the calling device. This results in the A leg being encrypted and > the B leg un-encrypted. However as soon as the key exchange initiates, the > A client errors and tears down the call. What's strange is that when I > initiate the call from the non-zrtp client to the zrtp client, the keys > exchange fine between server and client A, and 2-way audio is present. So > it's only when it's the dialling handset do I have the problem. > > > > I've then gone on and tested the same thing on the latest git, and the > problem goes away. So my question is, can anyone advise on what > changes/patches have been applied to the latest git that has resolved this > issue? I've tried to search the git tree, but I'll be honest I'm a bit > unfamiliar with how it works and I'm not sure that I'm searching properly. > > > > The ZRTP device is an iPhone running Groundwire, and it doesn't seem to > matter what codec I use. > > > > Thanks for the help! > > > > > From asilva at wirelessmundi.com Wed Jan 16 15:18:27 2013 From: asilva at wirelessmundi.com (Antonio Silva) Date: Wed, 16 Jan 2013 13:18:27 +0100 Subject: [Freeswitch-users] Freeswitch TLS and Yealink t26p In-Reply-To: <50E482F4.8060701@quentustech.com> References: <1356082510.23186.154.camel@marces.madrid.commsmundi.com> <1356090896.23186.170.camel@marces.madrid.commsmundi.com> <50E482F4.8060701@quentustech.com> Message-ID: <1358338707.5310.32.camel@vmmarces.vm.marces.com> Hi, Nice to now. I didn't have this problem, i could upload a custom certificate and with the option "Only Accept Trusted Certificates" to enabled. I had another issue with ldap navigation, i report to them, they were quite fast to reply, and there is a new firmware 6.70.0.120 that solves the problem. Best regards, Ant?nio Silva On Wed, 2013-01-02 at 10:56 -0800, William King wrote: > Another thing to be aware of is that there is an outstanding bug where > Yealinks are not able to load certain custom CA's because the time on > the phone is not being synced to NTP before the CA is validated and loaded. > > Once you changed the transport have you run into any new issues? > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 12/21/2012 03:54 AM, Antonio wrote: > > Answer to myself.... > > > > In the yealink configuration, in the account parameters, the "transport" > > must be force to TLS. > > > > I don't know why it just works.... Before i was using DNS-SRV, that > > should be the first option, yealink should have some issue here... i > > will report to them. > > > > > > Thanks, > > Ant?nio > > > > On Fri, 2012-12-21 at 10:35 +0100, Antonio wrote: > >> Hi, > >> > >> I'm trying to register a yealink with TLS, using my one certificates. > >> > >> I follow the wiki and In fs i have both agent.pem and cafile.pem . I > >> install in the phone the root certificate. > >> > >> But when i try to register, i have (tport log): > >> > >> > >> tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x808fb0) msg > >> 0x7fe9d0aa8180 from (udp/192.168.10.1:5060) has 340 bytes, veclen = 1 > >> tport.c:3004 tport_deliver() tport_deliver(0x808fb0): msg > >> 0x7fe9d0aa8180 (340 bytes) from udp/192.168.10.23:5060/sip next=(nil) > >> tport.c:4202 tport_release() tport_release(0x808fb0): 0x7fe9d01142f0 > >> by 0x7fe9d025d920 with 0x7fe9d0aa8180 > >> tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fe9c802aad0): > >> events IN > >> tport.c:869 tport_alloc_secondary() > >> tport_alloc_secondary(0x7fe9c802aad0): new secondary tport 0x7fe9c03e8450 > >> tport_type_tls.c:603 tport_tls_accept() > >> tport_tls_accept(0x7fe9c03e8450): new connection from > >> tls/192.168.10.36:48754/sips > >> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events > >> NEGOTIATING > >> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events > >> NEGOTIATING > >> tport_tls.c:526 tls_post_connection_check() > >> tls_post_connection_check(0x7fe9c03e8450): Peer did not provide X.509 > >> Certificate. > >> > >> > >> > >> I could make it work and have a register in the tls profile when i > >> check on the phone the option in Security->Trusted Certificates: "Only > >> Accept Trusted Certificates: DISABLED". > >> Could it be some bug in the yealink, or I?m missing something in the > >> conf... > >> > >> Another question, is there any problem if i choose to use this > >> configuration... since is the phone that ignores the certificate and > >> the validation is done by the server and not by the client. > >> > >> Can you help me? > >> > >> Thanks, > >> Ant?nio > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > -- > > > > Un cordial saludo / Best regards, > > > > _________________________ > > > > Ant?nio Silva > > > > E-mail:asilva at wirelessmundi.com > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/7b0a262f/attachment-0001.html From ben122uk at gmail.com Wed Jan 16 15:30:54 2013 From: ben122uk at gmail.com (Ben) Date: Wed, 16 Jan 2013 12:30:54 -0000 Subject: [Freeswitch-users] ZRTP error on call setup when dialling client uses ZRTP In-Reply-To: <20130116161108.9cc0a0da5fb17db3dfd23023@yandex.ru> References: <022601cdf3d5$4c8e9a10$e5abce30$@gmail.com> <20130116161108.9cc0a0da5fb17db3dfd23023@yandex.ru> Message-ID: <024c01cdf3e5$550f7870$ff2e6950$@gmail.com> Hi, Do you mean proxy_media=true in the dialplan? If so, setting proxy-media mode in the dialplan means that Freeswitch does not negotiate keys with the client. This is not the desired setup. When both devices use ZRTP and proxy media is set, the keys exchange directly between the clients, this seems to work fine. It's when only one client uses ZRTP do I have the problem. Thanks. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of royj Sent: 16 January 2013 12:11 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] ZRTP error on call setup when dialling client uses ZRTP What is the proxy-media mode on both sides, in both cases. It matters. In my case I have experienced this On Wed, 16 Jan 2013 10:36:08 -0000 "Ben" wrote: > Hi all, > > > > My server is running the latest git stable build of Freeswitch, all > config is default and server/clients are on simple LAN setup. I'm > having an issue when using ZRTP on one client only, specifically the dialling client. > > > > The calling device has ZRTP enabled, and the called device does not. > In this scenario, the FS server should act as a man in the middle and > exchange keys with the calling device. This results in the A leg > being encrypted and the B leg un-encrypted. However as soon as the > key exchange initiates, the A client errors and tears down the call. > What's strange is that when I initiate the call from the non-zrtp > client to the zrtp client, the keys exchange fine between server and > client A, and 2-way audio is present. So it's only when it's the dialling handset do I have the problem. > > > > I've then gone on and tested the same thing on the latest git, and the > problem goes away. So my question is, can anyone advise on what > changes/patches have been applied to the latest git that has resolved > this issue? I've tried to search the git tree, but I'll be honest I'm > a bit unfamiliar with how it works and I'm not sure that I'm searching properly. > > > > The ZRTP device is an iPhone running Groundwire, and it doesn't seem > to matter what codec I use. > > > > Thanks for the help! > > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From cal.leeming at simplicitymedialtd.co.uk Wed Jan 16 15:59:08 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 16 Jan 2013 12:59:08 +0000 Subject: [Freeswitch-users] Issue - playing mp4 video files In-Reply-To: References: Message-ID: On Wed, Jan 16, 2013 at 12:08 PM, Muqtheear S wrote: > > 1. Please tell me how to convert mpeg4-generic encoded audio data to PCMU > encoded audio data in .mp4 file. I tried with ffmpeg, but couldn't succeed. > This is pretty far off topic - there are hundreds of articles out there explaining this, use Google. > 2. If you have any mp4 file with PCMU sound track, please share with me. > I don't understand, why would you need someone to send you an mp4 file with PCMU sound track? What exactly are you trying to achieve? > 3. Does freeswitch comes with mpeg4-generic codec support? > A quick Google search shows this; http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/085558.html http://wiki.freeswitch.org/wiki/Specsheet https://www.google.co.uk/search?q=freeswitch+mp4 > > > Thanks & Regards, > Muqtheear.S > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/b4de54e3/attachment.html From gvvsubhashkumar at gmail.com Wed Jan 16 15:59:13 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Wed, 16 Jan 2013 18:29:13 +0530 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel Message-ID: Hi all, I have a freeswitch configured with freetdm and wanpipe.My problem is with dtmf. When calling from an external pstn number to freeswitch, dtmf seems to be detected properly and enqueued it in the channel consistently. But when call came to same channel again, dtmf which is stored in the buffer for the previous call is used in the current call.So when i am done with previous call i want to dequeue the dtmf in the channel .So the problem seems to be something I've misconfigured with the Sangoma card. Can anyone point me in the right direction? Thanks, Subhash. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/8717aee5/attachment.html From royj at yandex.ru Wed Jan 16 16:19:18 2013 From: royj at yandex.ru (royj) Date: Wed, 16 Jan 2013 17:19:18 +0400 Subject: [Freeswitch-users] ZRTP error on call setup when dialling client uses ZRTP In-Reply-To: <024c01cdf3e5$550f7870$ff2e6950$@gmail.com> References: <022601cdf3d5$4c8e9a10$e5abce30$@gmail.com> <20130116161108.9cc0a0da5fb17db3dfd23023@yandex.ru> <024c01cdf3e5$550f7870$ff2e6950$@gmail.com> Message-ID: <20130116171918.77f726c39966d1f2403be034@yandex.ru> What about "full processing media mode" (without any options of proxying media) on the client, which supports zrtp. May be useful to provide a description of the configuration On Wed, 16 Jan 2013 12:30:54 -0000 "Ben" wrote: > Hi, > > Do you mean proxy_media=true in the dialplan? If so, setting proxy-media > mode in the dialplan means that Freeswitch does not negotiate keys with the > client. This is not the desired setup. > > When both devices use ZRTP and proxy media is set, the keys exchange > directly between the clients, this seems to work fine. It's when only one > client uses ZRTP do I have the problem. > > Thanks. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of royj > Sent: 16 January 2013 12:11 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] ZRTP error on call setup when dialling > client uses ZRTP > > > What is the proxy-media mode on both sides, in both cases. It matters. In my > case I have experienced this > > On Wed, 16 Jan 2013 10:36:08 -0000 > "Ben" wrote: > > > Hi all, > > > > > > > > My server is running the latest git stable build of Freeswitch, all > > config is default and server/clients are on simple LAN setup. I'm > > having an issue when using ZRTP on one client only, specifically the > dialling client. > > > > > > > > The calling device has ZRTP enabled, and the called device does not. > > In this scenario, the FS server should act as a man in the middle and > > exchange keys with the calling device. This results in the A leg > > being encrypted and the B leg un-encrypted. However as soon as the > > key exchange initiates, the A client errors and tears down the call. > > What's strange is that when I initiate the call from the non-zrtp > > client to the zrtp client, the keys exchange fine between server and > > client A, and 2-way audio is present. So it's only when it's the dialling > handset do I have the problem. > > > > > > > > I've then gone on and tested the same thing on the latest git, and the > > problem goes away. So my question is, can anyone advise on what > > changes/patches have been applied to the latest git that has resolved > > this issue? I've tried to search the git tree, but I'll be honest I'm > > a bit unfamiliar with how it works and I'm not sure that I'm searching > properly. > > > > > > > > The ZRTP device is an iPhone running Groundwire, and it doesn't seem > > to matter what codec I use. > > > > > > > > Thanks for the help! > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From muqtheear.s at gmail.com Wed Jan 16 16:33:06 2013 From: muqtheear.s at gmail.com (Muqtheear S) Date: Wed, 16 Jan 2013 19:03:06 +0530 Subject: [Freeswitch-users] Issue - playing mp4 video files In-Reply-To: References: Message-ID: Thanks for the reply. Currently I am facing this below error. How should I proceed? mod_mp4 module is looking for Hint tracks in the .mp4 file. "Hint tracks are a series of instructions in a mp4 container file that tell a server how to transmit packets." To create hint tracks for a mp4 file is fairly simple with the mp4creator program, we just executed the: mp4creator -hint= [-p ] As we know .mp4 has both video+audio data in it. We could also find audio and video encoding information of a .mp4 file as shown below. #mp4info minnale.mp4 mp4info version 1.5.0.1 minnale.mp4: Track Type Info 1 video MPEG-4 Simple @ L1, 97.833 secs, 136 kbps, 176x144 @ 12.000041 fps 2 audio MPEG-4 AAC LC, 97.568 secs, 64 kbps, 32000 Hz 3 hint Payload MP4V-ES for track 1 4 hint Payload mpeg4-generic for track 2 Metadata Tool: mp4creator 1.6 Now we are getting "Invalid codec mpeg4-generic!" error. Please guide me. Do we need to have mpeg4-generic codec in Freeswitch server? Thanks & Regards, Muqtheear.S On Wed, Jan 16, 2013 at 6:29 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > > > On Wed, Jan 16, 2013 at 12:08 PM, Muqtheear S wrote: > >> >> 1. Please tell me how to convert mpeg4-generic encoded audio data to PCMU >> encoded audio data in .mp4 file. I tried with ffmpeg, but couldn't succeed. >> > > This is pretty far off topic - there are hundreds of articles out there > explaining this, use Google. > > >> 2. If you have any mp4 file with PCMU sound track, please share with me. >> > > I don't understand, why would you need someone to send you an mp4 file > with PCMU sound track? What exactly are you trying to achieve? > > >> 3. Does freeswitch comes with mpeg4-generic codec support? >> > > A quick Google search shows this; > > > http://lists.freeswitch.org/pipermail/freeswitch-users/2012-July/085558.html > http://wiki.freeswitch.org/wiki/Specsheet > https://www.google.co.uk/search?q=freeswitch+mp4 > > > >> >> >> Thanks & Regards, >> Muqtheear.S >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/ffde51c9/attachment-0001.html From ben122uk at gmail.com Wed Jan 16 16:38:14 2013 From: ben122uk at gmail.com (Ben) Date: Wed, 16 Jan 2013 13:38:14 -0000 Subject: [Freeswitch-users] ZRTP error on call setup when dialling client uses ZRTP In-Reply-To: <20130116171918.77f726c39966d1f2403be034@yandex.ru> References: <022601cdf3d5$4c8e9a10$e5abce30$@gmail.com> <20130116161108.9cc0a0da5fb17db3dfd23023@yandex.ru> <024c01cdf3e5$550f7870$ff2e6950$@gmail.com> <20130116171918.77f726c39966d1f2403be034@yandex.ru> Message-ID: <025901cdf3ee$bd4341c0$37c9c540$@gmail.com> Configuration is the default config that comes with installing Freeswitch, which means that neither proxy media or bypass media are set and the audio is being processed by Freeswitch. As mentioned below, the setup is a simple LAN setup with clients and server on the same LAN. I have no advanced config in the dialplan, all the user accounts are default, so it's just the stock Freeswitch build. Thanks. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of royj Sent: 16 January 2013 13:19 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] ZRTP error on call setup when dialling client uses ZRTP What about "full processing media mode" (without any options of proxying media) on the client, which supports zrtp. May be useful to provide a description of the configuration On Wed, 16 Jan 2013 12:30:54 -0000 "Ben" wrote: > Hi, > > Do you mean proxy_media=true in the dialplan? If so, setting > proxy-media mode in the dialplan means that Freeswitch does not > negotiate keys with the client. This is not the desired setup. > > When both devices use ZRTP and proxy media is set, the keys exchange > directly between the clients, this seems to work fine. It's when only > one client uses ZRTP do I have the problem. > > Thanks. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > royj > Sent: 16 January 2013 12:11 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] ZRTP error on call setup when dialling > client uses ZRTP > > > What is the proxy-media mode on both sides, in both cases. It matters. > In my case I have experienced this > > On Wed, 16 Jan 2013 10:36:08 -0000 > "Ben" wrote: > > > Hi all, > > > > > > > > My server is running the latest git stable build of Freeswitch, all > > config is default and server/clients are on simple LAN setup. I'm > > having an issue when using ZRTP on one client only, specifically the > dialling client. > > > > > > > > The calling device has ZRTP enabled, and the called device does not. > > In this scenario, the FS server should act as a man in the middle > > and exchange keys with the calling device. This results in the A > > leg being encrypted and the B leg un-encrypted. However as soon as > > the key exchange initiates, the A client errors and tears down the call. > > What's strange is that when I initiate the call from the non-zrtp > > client to the zrtp client, the keys exchange fine between server and > > client A, and 2-way audio is present. So it's only when it's the > > dialling > handset do I have the problem. > > > > > > > > I've then gone on and tested the same thing on the latest git, and > > the problem goes away. So my question is, can anyone advise on what > > changes/patches have been applied to the latest git that has > > resolved this issue? I've tried to search the git tree, but I'll be > > honest I'm a bit unfamiliar with how it works and I'm not sure that > > I'm searching > properly. > > > > > > > > The ZRTP device is an iPhone running Groundwire, and it doesn't seem > > to matter what codec I use. > > > > > > > > Thanks for the help! > > > > > > > > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From royj at yandex.ru Wed Jan 16 16:49:34 2013 From: royj at yandex.ru (royj) Date: Wed, 16 Jan 2013 17:49:34 +0400 Subject: [Freeswitch-users] ZRTP error on call setup when dialling client uses ZRTP In-Reply-To: <025901cdf3ee$bd4341c0$37c9c540$@gmail.com> References: <022601cdf3d5$4c8e9a10$e5abce30$@gmail.com> <20130116161108.9cc0a0da5fb17db3dfd23023@yandex.ru> <024c01cdf3e5$550f7870$ff2e6950$@gmail.com> <20130116171918.77f726c39966d1f2403be034@yandex.ru> <025901cdf3ee$bd4341c0$37c9c540$@gmail.com> Message-ID: <20130116174934.21f4008d527c2bf814b04574@yandex.ru> Well then debug and developer will help in this case, I think On Wed, 16 Jan 2013 13:38:14 -0000 "Ben" wrote: > Configuration is the default config that comes with installing Freeswitch, > which means that neither proxy media or bypass media are set and the audio > is being processed by Freeswitch. As mentioned below, the setup is a simple > LAN setup with clients and server on the same LAN. I have no advanced > config in the dialplan, all the user accounts are default, so it's just the > stock Freeswitch build. > > Thanks. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of royj > Sent: 16 January 2013 13:19 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] ZRTP error on call setup when dialling > client uses ZRTP > > > What about "full processing media mode" (without any options of proxying > media) on the client, which supports zrtp. May be useful to provide a > description of the configuration > > On Wed, 16 Jan 2013 12:30:54 -0000 > "Ben" wrote: > > > Hi, > > > > Do you mean proxy_media=true in the dialplan? If so, setting > > proxy-media mode in the dialplan means that Freeswitch does not > > negotiate keys with the client. This is not the desired setup. > > > > When both devices use ZRTP and proxy media is set, the keys exchange > > directly between the clients, this seems to work fine. It's when only > > one client uses ZRTP do I have the problem. > > > > Thanks. > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > royj > > Sent: 16 January 2013 12:11 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] ZRTP error on call setup when dialling > > client uses ZRTP > > > > > > What is the proxy-media mode on both sides, in both cases. It matters. > > In my case I have experienced this > > > > On Wed, 16 Jan 2013 10:36:08 -0000 > > "Ben" wrote: > > > > > Hi all, > > > > > > > > > > > > My server is running the latest git stable build of Freeswitch, all > > > config is default and server/clients are on simple LAN setup. I'm > > > having an issue when using ZRTP on one client only, specifically the > > dialling client. > > > > > > > > > > > > The calling device has ZRTP enabled, and the called device does not. > > > In this scenario, the FS server should act as a man in the middle > > > and exchange keys with the calling device. This results in the A > > > leg being encrypted and the B leg un-encrypted. However as soon as > > > the key exchange initiates, the A client errors and tears down the call. > > > What's strange is that when I initiate the call from the non-zrtp > > > client to the zrtp client, the keys exchange fine between server and > > > client A, and 2-way audio is present. So it's only when it's the > > > dialling > > handset do I have the problem. > > > > > > > > > > > > I've then gone on and tested the same thing on the latest git, and > > > the problem goes away. So my question is, can anyone advise on what > > > changes/patches have been applied to the latest git that has > > > resolved this issue? I've tried to search the git tree, but I'll be > > > honest I'm a bit unfamiliar with how it works and I'm not sure that > > > I'm searching > > properly. > > > > > > > > > > > > The ZRTP device is an iPhone running Groundwire, and it doesn't seem > > > to matter what codec I use. > > > > > > > > > > > > Thanks for the help! > > > > > > > > > > > > > > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Wed Jan 16 18:06:16 2013 From: dujinfang at gmail.com (Seven Du) Date: Wed, 16 Jan 2013 23:06:16 +0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: <20130116025242.83fde4e4@mail.tritonwest.net> References: <20130116025242.83fde4e4@mail.tritonwest.net> Message-ID: <1564E4488B8946BB80621FD70058A468@gmail.com> Hi Dave, Thanks for this info. Can you explain a little more detail about the TCP pipe? I can see SpanDSP create a /dev/FS0 on linux, but how to pipe a TCP connection to linux on Windows? Does SpanDSP also listen on a TCP port? Or need other software to map a TCP port to /dev/FS0? Thanks. On Wednesday, January 16, 2013 at 10:52 AM, Dave R. Kompel wrote: > In FreeSWITCH, mod_spandsp, will also emulate fax modems. Com0Com is like a PTY driver for windows. When you set modem_count in spandsp.conf.xml to a count other then 0, the windows implmentation starts at COM3. So to get T.38 emulated fax modems, you don't need anything other then FreeSWITCH. > > Another option would be on the Linux box, enable SPANDSP modems, and use com0com on windows to backup com ports to raw TCP connections. It supports both. That way it would talk over a TCP pipe to an emulated modem on your other FreeSWITCH box. > > --Dave > > > From: Steven Schoch [mailto:schoch+freeswitch.org at xwin32.com] > > To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > > Sent: Tue, 15 Jan 2013 16:39:15 -0800 > > Subject: Re: [Freeswitch-users] FAX problems > > > > My whole plan here is to replace the phone system currently in the office (a NexPath system, a really nice PC-chassis pbx system from the 1990's) to a more modern VoIP system, and git rid of all our POTS lines. But I need a simple solution for the users who send FAXes from their Windows systems. > > > > On Tue, Jan 15, 2013 at 10:48 AM, Dave R. Kompel wrote: > > > It seems that you may be debugging way too many things at once. Do you know much about the FAXBACK.COM (http://faxback.com/) internals? > > > > Yes, that's probably correct. No, I don't know much about faxback.com (http://faxback.com/). > > > > However, I just tried a different one, www.t38faxvoip.com (http://www.t38faxvoip.com/). It's a much more involved Windows installation, but it came up with the same symptoms. This is leading me to believe that the problem is more likely my FS configuration. > > > > > Just for a test you know you can bypass it. FreeSWITCH will work just fine on a windows box, using the FAXModem stuff, and com0com driver. > > > > That's not my objective. I'm still early in the learning process for FreeSwitch, and I'd like to keep it to one installation, on an OS that I can understand (Linux). > > > > > That's what I use to have the M$ fax service talk to, and it works great. No hardware, and FS doing T38 right out of the Fax Service Server. > > > > Let me see if I understand: MS Fax is talking through a virtual serial cable (com0com) to what it thinks is a FAX modem. On the other side of that virtual serial cable is something that takes the serial FAX data, translates it to T38, and injects it into FS. I'm a little unclear on the last 2 pieces, because it still sounds like a black box to me. What's "Fax Service Server"? > > > > > > On Mon, Jan 14, 2013 at 5:12 PM, Michael Collins wrote: > > > Also, are you routing the call to the txfax app? > > > > I'm not. I assumed that the purpose of txfax is to encode a TIFF file into a T.38 stream, and my source is already T.38. Am I missing something? > > > > -- > > Steve > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/e750c344/attachment.html From raimund.sacherer at logitravel.com Wed Jan 16 18:10:06 2013 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Wed, 16 Jan 2013 16:10:06 +0100 (CET) Subject: [Freeswitch-users] Kernel Hyperthreading ON or OFF? In-Reply-To: <5117750.7628.1358348784453.JavaMail.javamailuser@localhost> Message-ID: <12052758.7693.1358349000791.JavaMail.javamailuser@localhost> Hi, we have a new Freeswitch cluster with Dell R410. It has 2 CPUs (Xeon E5620, QuadCore. it's 8 cores withouth HT and 16 cores with HT. I googled with hyperthreading and freeswitch but some say turn it on it will give you a boost, others say turn it off, you will gain performance. Does anyone know if it's better to leave it on or turn it off? And some small explanation? I would be very gratefull for your help, best regards Raimund -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/f7367a36/attachment-0001.html From kris at kriskinc.com Wed Jan 16 18:27:58 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 16 Jan 2013 10:27:58 -0500 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <7FA4A571-7A65-4493-84A4-90052B9CADC8@edvina.net> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <7FA4A571-7A65-4493-84A4-90052B9CADC8@edvina.net> Message-ID: +1! On Wed, Jan 16, 2013 at 3:09 AM, Olle E. Johansson wrote: > > 16 jan 2013 kl. 05:02 skrev "support at ecn.net.au" : > > We're quite new to Freeswitch and are in the process of migrating from > OpenSer (as an SBC) to Freeswitch. > > It surprises me that you compare two very different products and migrate > from one to the other. OpenSER (or Kamailio.org which is the new name) and > FreeSwitch work very well together. FreeSwitch is a b2bua that handles the > media and Kamailio kan handle the SIP side and add a layer of protection > with modules like Pike that give DoS attack prevention, the SIP pipes for > QoS and much more. Don't migrate, use both together! > > /O > > Kamailio marketing dept :-) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From lists at kavun.ch Wed Jan 16 18:40:14 2013 From: lists at kavun.ch (Emrah) Date: Wed, 16 Jan 2013 10:40:14 -0500 Subject: [Freeswitch-users] Freeswitch TLS and Yealink t26p In-Reply-To: <1358338707.5310.32.camel@vmmarces.vm.marces.com> References: <1356082510.23186.154.camel@marces.madrid.commsmundi.com> <1356090896.23186.170.camel@marces.madrid.commsmundi.com> <50E482F4.8060701@quentustech.com> <1358338707.5310.32.camel@vmmarces.vm.marces.com> Message-ID: <6C4DC695-57FA-4CA8-9332-CC19B974B85A@kavun.ch> Sorry to hijack the thread, but where do you report bugs / feature requests? I haven't been able to reach any knowledgeable person via their support email. Cheers and thank you On Jan 16, 2013, at 7:18 AM, Antonio Silva wrote: > Hi, > > Nice to now. > I didn't have this problem, i could upload a custom certificate and with the option "Only Accept Trusted Certificates" to enabled. > > I had another issue with ldap navigation, i report to them, they were quite fast to reply, and there is a new firmware 6.70.0.120 that solves the problem. > > Best regards, > > Ant?nio Silva > > > On Wed, 2013-01-02 at 10:56 -0800, William King wrote: >> Another thing to be aware of is that there is an outstanding bug where >> Yealinks are not able to load certain custom CA's because the time on >> the phone is not being synced to NTP before the CA is validated and loaded. >> >> Once you changed the transport have you run into any new issues? >> >> William King >> Senior Engineer >> Quentus Technologies, INC >> 1037 NE 65th St Suite 273 >> Seattle, WA 98115 >> Main: (877) 211-9337 >> Office: (206) 388-4772 >> Cell: (253) 686-5518 >> >> william.king at quentustech.com >> >> >> On 12/21/2012 03:54 AM, Antonio wrote: >> > Answer to myself.... >> > >> > In the yealink configuration, in the account parameters, the "transport" >> > must be force to TLS. >> > >> > I don't know why it just works.... Before i was using DNS-SRV, that >> > should be the first option, yealink should have some issue here... i >> > will report to them. >> > >> > >> > Thanks, >> > Ant?nio >> > >> > On Fri, 2012-12-21 at 10:35 +0100, Antonio wrote: >> >> Hi, >> >> >> >> I'm trying to register a yealink with TLS, using my one certificates. >> >> >> >> I follow the wiki and In fs i have both agent.pem and cafile.pem . I >> >> install in the phone the root certificate. >> >> >> >> But when i try to register, i have (tport log): >> >> >> >> >> >> tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x808fb0) msg >> >> 0x7fe9d0aa8180 from (udp/192.168.10.1:5060) has 340 bytes, veclen = 1 >> >> tport.c:3004 tport_deliver() tport_deliver(0x808fb0): msg >> >> 0x7fe9d0aa8180 (340 bytes) from udp/192.168.10.23:5060/sip next=(nil) >> >> tport.c:4202 tport_release() tport_release(0x808fb0): 0x7fe9d01142f0 >> >> by 0x7fe9d025d920 with 0x7fe9d0aa8180 >> >> tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fe9c802aad0): >> >> events IN >> >> tport.c:869 tport_alloc_secondary() >> >> tport_alloc_secondary(0x7fe9c802aad0): new secondary tport 0x7fe9c03e8450 >> >> tport_type_tls.c:603 tport_tls_accept() >> >> tport_tls_accept(0x7fe9c03e8450): new connection from >> >> tls/192.168.10.36:48754/sips >> >> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events >> >> NEGOTIATING >> >> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events >> >> NEGOTIATING >> >> tport_tls.c:526 tls_post_connection_check() >> >> tls_post_connection_check(0x7fe9c03e8450): Peer did not provide X.509 >> >> Certificate. >> >> >> >> >> >> >> >> I could make it work and have a register in the tls profile when i >> >> check on the phone the option in Security->Trusted Certificates: "Only >> >> Accept Trusted Certificates: DISABLED". >> >> Could it be some bug in the yealink, or I?m missing something in the >> >> conf... >> >> >> >> Another question, is there any problem if i choose to use this >> >> configuration... since is the phone that ignores the certificate and >> >> the validation is done by the server and not by the client. >> >> >> >> Can you help me? >> >> >> >> Thanks, >> >> Ant?nio >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> >> consulting at freeswitch.org > > >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org >> >> >> >> http://wiki.freeswitch.org >> >> >> >> http://www.cluecon.com >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org > > >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> >> http://www.freeswitch.org >> >> > >> > -- >> > >> > Un cordial saludo / Best regards, >> > >> > _________________________ >> > >> > Ant?nio Silva >> > >> > E-mail: >> asilva at wirelessmundi.com > > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > >> consulting at freeswitch.org >> >> > >> http://www.freeswitchsolutions.com >> >> > >> > >> > >> >> >> > >> > Official FreeSWITCH Sites >> > >> http://www.freeswitch.org >> >> > >> http://wiki.freeswitch.org >> >> > >> http://www.cluecon.com >> >> > >> > FreeSWITCH-users mailing list >> > >> FreeSWITCH-users at lists.freeswitch.org >> >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lists at kavun.ch Wed Jan 16 18:41:44 2013 From: lists at kavun.ch (Emrah) Date: Wed, 16 Jan 2013 10:41:44 -0500 Subject: [Freeswitch-users] conference-flags In-Reply-To: <68031E41-1E74-49C5-AC80-2ABFCBB1DB52@kavun.ch> References: <68031E41-1E74-49C5-AC80-2ABFCBB1DB52@kavun.ch> Message-ID: <07101B0D-F9A9-4845-A9B8-5F591A9F8949@kavun.ch> Hi guys, If any of you know the answer to this I would be very grateful if you could take a minute and post it on the Wiki or reply to my email. Thanks! On Jan 11, 2013, at 2:28 PM, Emrah wrote: > Hi all, > > Where can I find some explanations on how to use conference-flags? > > How do you use video-bridge and video-floor and what do they do? > > Thanks for any pointers. > > E From jmesquita at freeswitch.org Wed Jan 16 18:49:16 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 16 Jan 2013 12:49:16 -0300 Subject: [Freeswitch-users] FAX problems In-Reply-To: <1564E4488B8946BB80621FD70058A468@gmail.com> References: <20130116025242.83fde4e4@mail.tritonwest.net> <1564E4488B8946BB80621FD70058A468@gmail.com> Message-ID: >From what I read, you need com0com compiled on Linux as well. So you do COM->IP on Windows and IP->TTY on Linux. Jo?o Mesquita FreeSWITCH? Solutions On Wed, Jan 16, 2013 at 1:06 PM, Seven Du wrote: > Hi Dave, > > Thanks for this info. Can you explain a little more detail about the TCP > pipe? I can see SpanDSP create a /dev/FS0 on linux, but how to pipe a TCP > connection to linux on Windows? Does SpanDSP also listen on a TCP port? Or > need other software to map a TCP port to /dev/FS0? > > Thanks. > > On Wednesday, January 16, 2013 at 10:52 AM, Dave R. Kompel wrote: > > In FreeSWITCH, mod_spandsp, will also emulate fax modems. Com0Com is > like a PTY driver for windows. When you set modem_count in spandsp.conf.xml > to a count other then 0, the windows implmentation starts at COM3. So to > get T.38 emulated fax modems, you don't need anything other then > FreeSWITCH. > > Another option would be on the Linux box, enable SPANDSP modems, and use > com0com on windows to backup com ports to raw TCP connections. It supports > both. That way it would talk over a TCP pipe to an emulated modem on your > other FreeSWITCH box. > > --Dave > > ------------------------------ > *From:* Steven Schoch [mailto:schoch+freeswitch.org at xwin32.com > ] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org > ] > *Sent:* Tue, 15 Jan 2013 16:39:15 -0800 > *Subject:* Re: [Freeswitch-users] FAX problems > > My whole plan here is to replace the phone system currently in the > office (a NexPath system, a really nice PC-chassis pbx system from the > 1990's) to a more modern VoIP system, and git rid of all our POTS lines. > But I need a simple solution for the users who send FAXes from their > Windows systems. > > On Tue, Jan 15, 2013 at 10:48 AM, Dave R. Kompel wrote: > > ** > It seems that you may be debugging way too many things at once. Do you > know much about the FAXBACK.COM internals? > > > Yes, that's probably correct. No, I don't know much about faxback.com. > > However, I just tried a different one, www.t38faxvoip.com. It's a much > more involved Windows installation, but it came up with the same symptoms. > This is leading me to believe that the problem is more likely my FS > configuration. > > Just for a test you know you can bypass it. FreeSWITCH will work just > fine on a windows box, using the FAXModem stuff, and com0com driver. > > > That's not my objective. I'm still early in the learning process for > FreeSwitch, and I'd like to keep it to one installation, on an OS that I > can understand (Linux). > > That's what I use to have the M$ fax service talk to, and it works > great. No hardware, and FS doing T38 right out of the Fax Service Server. > > > Let me see if I understand: MS Fax is talking through a virtual serial > cable (com0com) to what it thinks is a FAX modem. On the other side of > that virtual serial cable is something that takes the serial FAX data, > translates it to T38, and injects it into FS. I'm a little unclear on the > last 2 pieces, because it still sounds like a black box to me. What's "Fax > Service Server"? > > > On Mon, Jan 14, 2013 at 5:12 PM, Michael Collins > wrote: > > Also, are you routing the call to the txfax app? > > > I'm not. I assumed that the purpose of txfax is to encode a TIFF file > into a T.38 stream, and my source is already T.38. Am I missing something? > > -- > Steve > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/c3655ae3/attachment-0001.html From benoit.raymond at amiconcept.com Wed Jan 16 19:21:33 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Wed, 16 Jan 2013 11:21:33 -0500 Subject: [Freeswitch-users] need help on call recording In-Reply-To: References: Message-ID: <088a01cdf405$8f122fc0$ad368f40$@amiconcept.com> Hi, It is most likely recorded but since you have RECORD_STEREO=true, the audio is ... stereo! If you load your wav in an editor (like Goldwave), that should be evident. So drop the RECORD_STEREO parameter or set it to false (this is the default anyway) and check if that works. Benoit Raymond AMI Concept Inc. Business VoIP Solutions / Web Design Phone: (450) 553-1231 http://www.amiconcept.com -----Message d'origine----- De?: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de haloha Envoy??: 16 janvier 2013 03:50 ??: FreeSWITCH Users Help Objet?: [Freeswitch-users] need help on call recording hi all i try to record all calls to/from local extension so my dialplan FS only records the caller voicem FS doesnt record the called voice How do i record the entire calls(caller and called) Thank you Ha` _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Wed Jan 16 19:23:05 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Jan 2013 10:23:05 -0600 Subject: [Freeswitch-users] Mod_Conference Improvement (Denoise) In-Reply-To: <5274F7F1-8BC9-4721-B98F-A163495CD730@jerris.com> References: <5274F7F1-8BC9-4721-B98F-A163495CD730@jerris.com> Message-ID: There is also the preprocess app which uses a media bug in the core for a variety of things but is incomplete it's using a bit of code from speex so we can certainly improve that as well. On Tue, Jan 15, 2013 at 8:58 PM, Michael Jerris wrote: > If you wanted to implement something like this you could do so as a media > bug, it does not need to touch mod_conference in any way. Take a look at > mod_ladspa for an example of a similar module that modifies audio using a > media bug. Give it a try and see if you can get something working. > > Mike > > On Jan 15, 2013, at 8:03 PM, Michael Collins wrote: > > While I don't have a problem with the concept of noise filtering I have to > point out that each member of the conference can have his or her own noise > level setting. From fs_cli: > > conference energy [] > > You only need to apply it to the person who is in the noisy location. In > fact, in the example configuration the user can dial 9 to increase the > energy threshold (for when he's in a noisy environment) or press 7 to > decrease the threshold. Pressing 8 will reset the energy to normal, which I > believe is 200. > > -MC > > On Tue, Jan 15, 2013 at 4:24 PM, Usama Zaidi wrote: > >> Hi, >> >> I talked about this a while back on the channel, the way how energy-level >> is implemented in mod_conference isn't the right way to handle noise in my >> opinion because the participants that get bridged would have a different >> noise floor depending on where they are located, so if I set the value of >> energy-level too high to accommodate one participant, any other participant >> in a quite sitting would never get bridged, I proposed we should tackle >> this the way Asterisk does using a denoise dialplan function ( >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE), >> we can use libav for noise filtering on the muxed audio. I'm willing to >> contribute if one of the devs would help me out regarding implementation. >> Any comments? >> >> -Regards >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/8bf769c6/attachment.html From support at ecn.net.au Wed Jan 16 15:55:52 2013 From: support at ecn.net.au (support at ecn.net.au) Date: Wed, 16 Jan 2013 22:55:52 +1000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting>, Message-ID: <8A80C99638A5E24E83696EAE33A9B6B968B53027EF@twbexmbx001.ecn.hosting> Hi This is what we now have, however there is an interesting side effect (major issues actally) we're getting when using start_dtmf on the aleg from the "dodgy" telco. the oddity is that when start_dtmf is executed prior to bridge some audio content from the back end PBX's (behind the Freeswitch SBC) does not seem to transmit back through to the caller (on the aleg of the freeswitch SBC (for example play back of a canned audio file from an IVR on an Asterisk PBX behind the SBC). In testing Freeswitch on version 1.2 (1.2.5 and 1.2.3) and 1.3 testing all versions have this issue, however on an old 1.0.6 legacy version that we compiled today - does not produce the same problem (it works perfectly); could it be a bug in the 1.2/3 freeswitch do you think? Or is it something we've missed? To reconfirm: context A (telco context) we have a sip profile (dtmf_mode=none), context B (to.pbx context) we have a sip profile (dtmf_mode=rfc2833) On the xml dial plan for context A we start_dtmf before we bridge the call from the telco to the backend pbx. When executing start_dtmf the system correctly transmits the dtmf digits to the backend pbx's however the by product (which we earlier thought was clipping the start of the call) is that some audio content (such as some audio file playback) doesn't get heard by the caller (dialed via the telco through the SBC to the pbx). Any help appreciated! Is there a change in how start_dtmf effects calls from the 1.2 and onward versions? Regards Mark ________________________________________ From: jay binks [jaybinks at gmail.com] Sent: Wednesday, 16 January 2013 7:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SBC In-band DTMF on your internal sip profile ( one facing PBX's ) set then note the dialplan context that super dodgey sip carrier ( the one doing inband only ) sends calls to, and use on calls from them. Id advise against in that location unless you REALLY want that. that should do what you want. On 16 January 2013 14:02, support at ecn.net.au > wrote: Hi All We're quite new to Freeswitch and are in the process of migrating from OpenSer (as an SBC) to Freeswitch. Mostly all is working well, except an oddity on DTMF. Our scenario: Telco/SIP Provider A is passing us calls using DTMF inband. We have a freeswitch configured as a SBC using 2 sip profiles (telco and internal) to topology hide and manage distribution of calls to the PBX servers located behind the SBC. The freeswitch will be handling up to a few hundred calls so we're trying to keep it lightweight. Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling customer needs. An example inbound call profile looks like this: > Initially when calling into the platform IVR type applications runinng on our PBX boxes would not work (you could hear the DTMF but the platform did not recognise the tones). We have had to add the appliation start_dtmf in order for Freeswitch to pass the DTMF to the Asterisk PBX behind the SBC. Interestingly on our OpenSer platform we just proxied the media (rtpproxy) with inband DTMF from the Telco and our PBX boxes recognised the inband DTMF tones on the PBX platforms and IVR type applications just worked. However under freeswitch if we don't start_dtmf before the bridge the backend PBX boxes don't recognise the DTMF inband (even though the tones are audible ie you can hear them on a call recording on the PBX). Have we missed something here? We would have thought with inband DTMF on non compressed codec (no transcoding) that the tones would just work with the media stream? We have confirmed both legs are PCMA and when using start_dtmf the first second of the call is clipped. Kind Regards, _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sincerely Jay From ptness at gmx.co.uk Wed Jan 16 11:51:00 2013 From: ptness at gmx.co.uk (peter ness) Date: Wed, 16 Jan 2013 09:51:00 +0100 Subject: [Freeswitch-users] compile error Message-ID: <20130116085100.259790@gmx.com> Can anybody advise why I am getting this error in compiling. This is a fresh download fron git and default modules.conf and ./configure Systen is amd x86_64-pc-linux-gnu make[2]: Leaving directory `/root/Desktop/freeswitch/libs/curl' make[1]: Leaving directory `/root/Desktop/freeswitch/libs/curl' which: no git in (/usr/local/sbin:/usr/sbin:/sbin:/usr/local/bin:/usr/bin:/bin:/usr/games:/usr/lib64/kde4/libexec:/usr/lib64/qt/bin) cat /root/Desktop/freeswitch/src/include/switch_cpp.h | perl /root/Desktop/freeswitch/build/strip.pl > /root/Desktop/freeswitch/src/include/switch_swigable_cpp.h make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /root/Desktop/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /root/Desktop/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /root/Desktop/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /root/Desktop/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /root/Desktop/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /root/Desktop/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; ?mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /root/Desktop/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /root/Desktop/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive quiet_libtool: compile: ?gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/root/Desktop/freeswitch/libs/curl/include -I/root/Desktop/freeswitch/src/include -I/root/Desktop/freeswitch/src/include -I/root/Desktop/freeswitch/libs/libteletone/src -I/root/Desktop/freeswitch/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -Ilibs/sofia-sip/libsofia-sip-ua/sdp/sofia-sip -Ilibs/sofia-sip/libsofia-sip-ua/su -g -O2 -pthread -D_REENTRANT -D_GNU_SOURCE -I/root/Desktop/freeswitch/libs/apr/include -I/root/Desktop/freeswitch/libs/apr-util/include -I/root/Desktop/freeswitch/libs/apr-util/xml/expat/lib -I/root/Desktop/freeswitch/libs/libtpl-1.5/src -I/root/Desktop/freeswitch/libs/stfu -I/root/Desktop/freeswitch/libs/sqlite -I/root/Desktop/freeswitch/libs/pcre -I/root/Desktop/freeswitch/libs/speex/include -Ilibs/speex/include -I/root/Desktop/freeswitch/libs/srtp/include -I/root/Desktop/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/root/Desktop/freeswitch/libs/spandsp/src -I/root/Desktop/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/local/include -I/root/Desktop/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/root/Desktop/freeswitch/libs/curl/include -I/root/Desktop/freeswitch/src/include -I/root/Desktop/freeswitch/src/include -I/root/Desktop/freeswitch/libs/libteletone/src -I/root/Desktop/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_apr.lo -MD -MP -MF .deps/libfreeswitch_la-switch_apr.Tpo -c src/switch_apr.c ?-fPIC -DPIC -o .libs/libfreeswitch_la-switch_apr.o In file included from /root/Desktop/freeswitch/libs/spandsp/src/spandsp.h:66:0, ? ? ? ? ? ? ? ? from ./src/include/private/switch_core_pvt.h:35, ? ? ? ? ? ? ? ? from src/switch_apr.c:37: /root/Desktop/freeswitch/libs/spandsp/src/spandsp/fir.h: In function 'fir16_create': /root/Desktop/freeswitch/libs/spandsp/src/spandsp/fir.h:94:5: error: implicit declaration of function 'rpl_malloc' [-Werror=implicit-function-declaration] /root/Desktop/freeswitch/libs/spandsp/src/spandsp/fir.h:94:25: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast] /root/Desktop/freeswitch/libs/spandsp/src/spandsp/fir.h: In function 'fir32_create': /root/Desktop/freeswitch/libs/spandsp/src/spandsp/fir.h:213:20: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast] /root/Desktop/freeswitch/libs/spandsp/src/spandsp/fir.h: In function 'fir_float_create': /root/Desktop/freeswitch/libs/spandsp/src/spandsp/fir.h:261:20: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast] src/switch_apr.c: In function 'switch_dir_open': src/switch_apr.c:558:22: error: initialization makes pointer from integer without a cast [-Werror] cc1: all warnings being treated as errors make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make: *** [all] Error 2 From support at ecn.net.au Wed Jan 16 16:15:54 2013 From: support at ecn.net.au (support at ecn.net.au) Date: Wed, 16 Jan 2013 23:15:54 +1000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <8A80C99638A5E24E83696EAE33A9B6B968B53027EF@twbexmbx001.ecn.hosting> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting>, , <8A80C99638A5E24E83696EAE33A9B6B968B53027EF@twbexmbx001.ecn.hosting> Message-ID: <8A80C99638A5E24E83696EAE33A9B6B968B53027F0@twbexmbx001.ecn.hosting> Following up on this. We've tested a few more configurations and it seems that when we exec start_dtmf it is the first perhaps half a second of any sound file that is played on a back end PBX seems to be clipped. If we download the same sound file and play it it is fine, if we use Freeswitch 1.0.6 (intstead of the stable version) it is also fine. an example sounds file is www.ecn.net.au/example.gsm - this file played off an Astiersk PBX befind the Freeswitch SBC plays, however the first 1/2 second or is not heard by the caller (calling through the telco, routed off the SBC and answered in the backend PBX). (if you listen the first part that says "please press 1 to" is not heard, but the rest of the file is. Is there some Freeswitch configuration that may cause this? Regards Mark ________________________________________ From: support at ecn.net.au Sent: Wednesday, 16 January 2013 10:55 PM To: FreeSWITCH Users Help Subject: RE: [Freeswitch-users] SBC In-band DTMF Hi This is what we now have, however there is an interesting side effect (major issues actally) we're getting when using start_dtmf on the aleg from the "dodgy" telco. the oddity is that when start_dtmf is executed prior to bridge some audio content from the back end PBX's (behind the Freeswitch SBC) does not seem to transmit back through to the caller (on the aleg of the freeswitch SBC (for example play back of a canned audio file from an IVR on an Asterisk PBX behind the SBC). In testing Freeswitch on version 1.2 (1.2.5 and 1.2.3) and 1.3 testing all versions have this issue, however on an old 1.0.6 legacy version that we compiled today - does not produce the same problem (it works perfectly); could it be a bug in the 1.2/3 freeswitch do you think? Or is it something we've missed? To reconfirm: context A (telco context) we have a sip profile (dtmf_mode=none), context B (to.pbx context) we have a sip profile (dtmf_mode=rfc2833) On the xml dial plan for context A we start_dtmf before we bridge the call from the telco to the backend pbx. When executing start_dtmf the system correctly transmits the dtmf digits to the backend pbx's however the by product (which we earlier thought was clipping the start of the call) is that some audio content (such as some audio file playback) doesn't get heard by the caller (dialed via the telco through the SBC to the pbx). Any help appreciated! Is there a change in how start_dtmf effects calls from the 1.2 and onward versions? Regards Mark ________________________________________ From: jay binks [jaybinks at gmail.com] Sent: Wednesday, 16 January 2013 7:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SBC In-band DTMF on your internal sip profile ( one facing PBX's ) set then note the dialplan context that super dodgey sip carrier ( the one doing inband only ) sends calls to, and use on calls from them. Id advise against in that location unless you REALLY want that. that should do what you want. On 16 January 2013 14:02, support at ecn.net.au > wrote: Hi All We're quite new to Freeswitch and are in the process of migrating from OpenSer (as an SBC) to Freeswitch. Mostly all is working well, except an oddity on DTMF. Our scenario: Telco/SIP Provider A is passing us calls using DTMF inband. We have a freeswitch configured as a SBC using 2 sip profiles (telco and internal) to topology hide and manage distribution of calls to the PBX servers located behind the SBC. The freeswitch will be handling up to a few hundred calls so we're trying to keep it lightweight. Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling customer needs. An example inbound call profile looks like this: > Initially when calling into the platform IVR type applications runinng on our PBX boxes would not work (you could hear the DTMF but the platform did not recognise the tones). We have had to add the appliation start_dtmf in order for Freeswitch to pass the DTMF to the Asterisk PBX behind the SBC. Interestingly on our OpenSer platform we just proxied the media (rtpproxy) with inband DTMF from the Telco and our PBX boxes recognised the inband DTMF tones on the PBX platforms and IVR type applications just worked. However under freeswitch if we don't start_dtmf before the bridge the backend PBX boxes don't recognise the DTMF inband (even though the tones are audible ie you can hear them on a call recording on the PBX). Have we missed something here? We would have thought with inband DTMF on non compressed codec (no transcoding) that the tones would just work with the media stream? We have confirmed both legs are PCMA and when using start_dtmf the first second of the call is clipped. Kind Regards, _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sincerely Jay From gassaad at emassembly.com Wed Jan 16 17:57:51 2013 From: gassaad at emassembly.com (George Assaad) Date: Wed, 16 Jan 2013 09:57:51 -0500 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: Message-ID: Sorry for the delay to get back to you. 1- App --> fax server 2- Create two extensions (one for inbound) another for (outbound) From within either extension you can send faxes "virtual fax" 3- point your create your outbound and inbound dialplans as normal but make it to point to those extensions. Good luck, George On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig wrote: > What did you do on the config side to get it to work? I've tried both an > Ata and a fax to email extension and they both fail in both directions. > > Robert > > On Jan 12, 2013, at 2:04 PM, "George Assaad" > wrote: > > I use Fusion successfully to send / receive fax. > The way it works: > ATA --->FaxServer. > After successful receipt, Fax Server---> DID where you > have dialled originally. > > Cheers, > > George > > > On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig wrote: > >> Did anyone see the issue or what I may have missed in the config? As >> mentioned earlier, I thought it would after a fresh install. Any help is >> appreciated. >> >> Robert >> >> ------------------------------ >> From: craig.robert at hotmail.com >> To: freeswitch-users at lists.freeswitch.org >> Date: Thu, 10 Jan 2013 13:43:09 -0700 >> >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Thanks for the quick response. Below is the capture from the log viewer. >> In this test, I am using a remote fax machine (line from Cox with physical >> machine) faxing into the "email server" where I created an extension for >> fax-to-email. Calling number is 1479-246-0547. >> >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 >> (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 >> sofia/external/+14792460547 at flowroute.com Standard DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/ >> +14792460547 at flowroute.com SOFIA DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 >> (sofia/external/+14792460547 at flowroute.com) State DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 >> (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> >> DOWN >> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close >> Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] >> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 >> (sofia/external/+14792460547 at flowroute.com) Ended >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 >> (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external >> entities >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 >> (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> >> CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 >> (sofia/external/+14792460547 at flowroute.com) State REPORTING going to >> sleep >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 >> sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: >> NORMAL_CLEARING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 >> (sofia/external/+14792460547 at flowroute.com) State REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 >> (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> >> CS_REPORTING >> >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup >> Command with no Session system(/usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php email= >> robert.l.craig at craignetwork.com extension=3150 >> name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after >> permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 >> caller_id_name='' caller_id_number=): >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 >> (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 >> sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: >> NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to >> sofia/external/+14792460547 at flowroute.com >> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel >> sofia/external/+14792460547 at flowroute.com hanging up, cause: >> NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 >> (sofia/external/+14792460547 at flowroute.com) State HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 >> sofia/external/+14792460547 at flowroute.com skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal >> sofia/external/+14792460547 at flowroute.com [KILL] >> 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup >> sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] >> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+ >> 14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP >> EXECUTE sofia/external/+14792460547 at flowroute.com hangup() >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 >> sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 >> ============================================================================== >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: >> 14400 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image >> resolution: 0x0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: >> 0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages >> transferred: 0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station >> id: SpanDSP Fax Ident >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station >> id: ATW-4792460547 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing >> not successful - result (48) Disconnected after permitted retries. >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 >> ============================================================================== >> 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port >> confirmed. >> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+ >> 14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ >> 66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ >> 66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+ >> 14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ >> 66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ >> 66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+ >> 14792460547 at flowroute.com Set rtp dtmf delay to 40 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf >> receive payload to 101 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send >> payload to 101 >> 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer >> spec [0] must be between 20 and 10000 >> 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer >> [soft] 160 bytes per 20ms >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP >> [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> >> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [ready][200] >> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxRateManagement:transferredTCF >> m=image 31632 udptl t38 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849876 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [completing][200] >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+ >> 14792460547 at flowroute.com Update Callee ID to "+14792460547" < >> +14792460547> >> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [calling][0] >> 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+ >> 14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec >> activation Success L16 >> 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 >> sofia/external/+14792460547 at flowroute.com Push codec L16:70 >> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec >> activation Success L16 20000 >> EXECUTE sofia/external/+14792460547 at flowroute.comrxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) >> 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done >> playing file silence_stream://2000 >> 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port >> confirmed. >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [ready][200] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec >> Activated L16 at 8000hz 1 channels 20ms >> EXECUTE sofia/external/+14792460547 at flowroute.complayback(silence_stream://2000) >> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET >> [last_fax]=[+14792460547-2013-01-10-13-39-26] >> EXECUTE sofia/external/+14792460547 at flowroute.comset(last_fax=+14792460547-2013-01-10-13-39-26) >> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.comset(fax_enable_t38_request=true) >> 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [fax_enable_t38]=[true] >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [completed][200] >> EXECUTE sofia/external/+14792460547 at flowroute.comset(fax_enable_t38=true) >> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel >> [sofia/external/+14792460547 at flowroute.com] has been answered >> 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+ >> 14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE >> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=sendrecv >> a=ptime:20 >> a=silenceSupp:off - - - - >> a=fmtp:101 0-16 >> a=rtpmap:101 telephone-event/8000 >> a=rtpmap:0 PCMU/8000 >> m=audio 27588 RTP/AVP 0 101 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP >> sofia/external/+14792460547 at flowroute.com: >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+ >> 14792460547 at flowroute.com Set rtp dtmf delay to 40 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf >> receive payload to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send >> payload to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer >> [soft] 160 bytes per 20ms >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP >> [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> >> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success >> [72.201.112.131]:[27588] >> EXECUTE sofia/external/+14792460547 at flowroute.com answer() >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php email= >> robert.l.craig at craignetwork.com extension=3150 name=${last_fax} >> messages='result: ${fax_result_text} sender:${fax_remote_station_id} >> pages:${fax_document_total_pages}' domain=192.168.4.15 >> caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] >> EXECUTE sofia/external/+14792460547 at flowroute.comset(api_hangup_hook=system /usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php email= >> robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} >> messages='result: \${fax_result_text} sender:\${fax_remote_station_id} >> pages:\${fax_document_total_pages}' domain=192.168.4.15 >> caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 >> -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, >> 10 Jan 2013 13:39:25 -0700) >> EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) >> EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-last_dial/+14792460547/3150) >> EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-spymap/ >> +14792460547/d276f012-5b65-11e2-beae-8564ff827a25) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [call_direction]=[local] >> EXECUTE sofia/external/+14792460547 at flowroute.comset(call_direction=local) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 >> sofia/external/+14792460547 at flowroute.com Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+ >> 14792460547 at flowroute.com SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 >> (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> >> CS_EXECUTE >> Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> playback(silence_stream://2000) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(fax_enable_t38_request=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(fax_enable_t38=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(api_hangup_hook=system /usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php email= >> robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} >> messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} >> pages:\\\${fax_document_total_pages}' domain=192.168.4.15 >> caller_id_name='\\\${caller_id_name}' >> caller_id_number=\\\${caller_id_number}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test >> Fax] destination_number(3150) =~ /^3150$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->Test Fax] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->vmain_user] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] >> destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->vmain] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->send_to_voicemail] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->extension-intercom] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->intercept-ext] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->call_return] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->call_privacy] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> hash(insert/${domain_name}-last_dial/global/${uuid}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition >> [global] >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->global] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [redial] destination_number(3150) =~ /^(redial|\*870)$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->redial] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->group-intercept] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action >> set(call_direction=local) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->call-direction] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->Flowroute.11d] continue=false >> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing >> +14792460547 <+14792460547>->3150 in context default >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 >> sofia/external/+14792460547 at flowroute.com Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+ >> 14792460547 at flowroute.com SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com) State ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep >> 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer >> sofia/external/+14792460547 at flowroute.com to XML[3150 at default] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+ >> 14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING >> EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML >> default) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT >> (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.comexport(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [outside_call]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 >> sofia/external/+14792460547 at flowroute.com Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+ >> 14792460547 at flowroute.com SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 >> (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> >> CS_EXECUTE >> Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 >> XML default) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (PASS) [Test_Number] context(public) =~ /public/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [public->Test_Number] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [public->call_debug] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(outside_call=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition >> [outside_call] >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [public->outside_call] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [public->unloop] continue=false >> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing >> +14792460547 <+14792460547>->15203039810 in context public >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 >> sofia/external/+14792460547 at flowroute.com Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+ >> 14792460547 at flowroute.com SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com) State ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+ >> 14792460547 at flowroute.com) Callstate Change DOWN -> RINGING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 >> (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+ >> 14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+ >> 14792460547 at flowroute.com SOFIA INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 >> (sofia/external/+14792460547 at flowroute.com) State INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+ >> 14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf >> send/recv payload to 101 >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 >> sofia/external/+14792460547 at flowroute.com Original read codec set to >> PCMU:0 >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec >> sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples >> 64000 bits >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare >> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >> >> a=ptime:20 >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:18 G729/8000 >> m=audio 31632 RTP/AVP 0 18 101 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849875 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [received][100] >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 >> (sofia/external/+14792460547 at flowroute.com) State NEW >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel >> sofia/external/+14792460547 at flowroute.com[d276f012-5b65-11e2-beae-8564ff827a25] >> >> Robert >> >> >> ------------------------------ >> Date: Thu, 10 Jan 2013 09:47:18 -0800 >> From: msc at freeswitch.org >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Can you pastebin a console debug log w/ sip trace of a call failing to >> negotiate t.38 correctly? That might help us figure out where to look. >> -MC >> >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: >> >> Just installed FusionPBX using the script built for a Centos 6 install. >> Everything works great in regards to calls, but my main goal of this >> project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have >> tried many things in media bypass, routing to a fax extension, using an >> ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me >> in the right direction as to where I should begin? I am guessing it isn't >> supposed to work right out of the box. I am using Flowroute as my provider >> right now and the have confirmed that they utilize T38 in both directions. >> Thanks! >> >> >> Robert >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official FreeSWITCH >> Sites http://www.freeswitch.org http://wiki.freeswitch.org >> http://www.cluecon.com FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official FreeSWITCH >> Sites http://www.freeswitch.org http://wiki.freeswitch.org >> http://www.cluecon.com FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/61d766df/attachment-0001.html From gassaad at emassembly.com Wed Jan 16 19:40:49 2013 From: gassaad at emassembly.com (George Assaad) Date: Wed, 16 Jan 2013 11:40:49 -0500 Subject: [Freeswitch-users] Redundant Dial Plan in JS In-Reply-To: <18af01cdf349$58bb6250$0a3226f0$@bizfocused.com> References: <0b1701cdf206$6a567930$3f036b90$@bizfocused.com> <18af01cdf349$58bb6250$0a3226f0$@bizfocused.com> Message-ID: If you use the vertical bar then you do not need to add a line for "continue_on_fail=true" It acts the same imo but not sure of the difference if there is any difference. On Tue, Jan 15, 2013 at 12:54 PM, Sean Devoy wrote: > Thanks Ken.**** > > ** ** > > I am left with a few questions though. If I use:**** > > **** > > ** ** > > What happens if the destination number is just plain old BUSY?**** > > ** ** > > Does the dialing user hear the fast-busy signal from ?primary? when their > route is defective and then hear the ring from secondary?**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice > *Sent:* Sunday, January 13, 2013 11:52 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Redundant Dial Plan in JS**** > > ** ** > > See > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover > > Or see using the bridge command in conjunction with the continue_on_fail > channel variable... > > This is pretty easy to do... Keep in mind that what good ol? ma bell does > is try one route and then if it fails, try to re-route around that... > That?s all the above is showing you how to do > > K > > > On 1/13/13 9:22 PM, "Sean Devoy" wrote:**** > > Hi, > > This question stems from the fact that I cannot find a VOIP Service > Provider (VSP) that can connect outgoing calls truly reliably. I have > customers who expect calls to go through just like good old ma-bell did > ?100% of the time barring natural disasters. You have to admit those guys > set the bar pretty high for us. > > I was using Voip Innovations with decent success, but my users complain > every 2 weeks or so and I send the trace to VI. The standard response is > always ?We have updated our route tables for that area.? I switched to > Vitelity where I was assured they have redundant paths and this could never > happen! I have only confirmed it happening twice in the last 3 weeks! Of > course one of those was during the demo of the new service I had just > installed for a new customer. > > So, I give up. My new approach is ?VSPs cannot deliver consistent highly > reliable service ? work around it!? > > I know I want a dial plan that will check the bridge status and redial > using a different gateway (VSP) if it failed (not busy or no answer ? > failure). I am pretty sure the best/easy approach is a JavaScript ?dialer? > script to handle this. > > I have looked in the wiki for such a sample. If it exists I have missed > it. So rather than ?re-inventing the wheel? I am reaching out to you all. > Does anyone have a dialing script that checks for failure and redials on > failure or just continues to voicemail on no answer? I will happily test > and tweak it and add it to the wiki if you have something to get me > started!!! If not I shall take the plunge and get started! My ideal > solution will dial, play a message and redial on failure, maybe 2 or three > times or play a failure message and email me on excess errors. I am > leaning toward ?Call congestion, rerouting. Please stand by? while the > new bridge is being attempted. > > Thanks, > Sean**** > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/25bd7470/attachment.html From craig.robert at hotmail.com Wed Jan 16 19:54:10 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Wed, 16 Jan 2013 09:54:10 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: , , , , , , Message-ID: OK, I've already tried creating one extension for inbound and set an email address. Of course, it failed and the errors are at the bottom of this email. What is the purpose of the outbound extension? Robert Craig Fort Huachuca, AZ 85613 520-226-9505 (Home) 760-583-8270 (Cell) Date: Wed, 16 Jan 2013 09:57:51 -0500 From: gassaad at emassembly.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Sorry for the delay to get back to you.1- App --> fax server2- Create two extensions (one for inbound) another for (outbound) From within either extension you can send faxes "virtual fax" 3- point your create your outbound and inbound dialplans as normal but make it to point to those extensions. Good luck, George On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig wrote: What did you do on the config side to get it to work? I've tried both an Ata and a fax to email extension and they both fail in both directions. Robert On Jan 12, 2013, at 2:04 PM, "George Assaad" wrote: I use Fusion successfully to send / receive fax.The way it works: ATA --->FaxServer.After successful receipt, Fax Server---> DID where you have dialled originally. Cheers, George On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig wrote: Did anyone see the issue or what I may have missed in the config? As mentioned earlier, I thought it would after a fresh install. Any help is appreciated. Robert From: craig.robert at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Thu, 10 Jan 2013 13:43:09 -0700 Subject: Re: [Freeswitch-users] T38 Configuration Thanks for the quick response. Below is the capture from the log viewer. In this test, I am using a remote fax machine (line from Cox with physical machine) faxing into the "email server" where I created an extension for fax-to-email. Calling number is 1479-246-0547. 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 sofia/external/+14792460547 at flowroute.com Standard DESTROY 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/+14792460547 at flowroute.com SOFIA DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> DOWN 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 (sofia/external/+14792460547 at flowroute.com) Ended 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external entities 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> CS_DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING going to sleep 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: NORMAL_CLEARING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> CS_REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup Command with no Session system(/usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 caller_id_name='' caller_id_number=): 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: NORMAL_CLEARING 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to sofia/external/+14792460547 at flowroute.com 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel sofia/external/+14792460547 at flowroute.com hanging up, cause: NORMAL_CLEARING 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_HANGUP 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 sofia/external/+14792460547 at flowroute.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal sofia/external/+14792460547 at flowroute.com [KILL] 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP EXECUTE sofia/external/+14792460547 at flowroute.com hangup() 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 14400 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station id: ATW-4792460547 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (48) Disconnected after permitted retries. 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer spec [0] must be between 20 and 10000 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxRateManagement:transferredTCF m=image 31632 udptl t38 t=0 0 c=IN IP4 66.42.121.5 s=- o=- 275426305 1357849876 IN IP4 66.42.121.5 v=0 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completing][200] 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+14792460547 at flowroute.com Update Callee ID to "+14792460547" <+14792460547> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [calling][0] 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec activation Success L16 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 sofia/external/+14792460547 at flowroute.com Push codec L16:70 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec activation Success L16 20000 EXECUTE sofia/external/+14792460547 at flowroute.com rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done playing file silence_stream://2000 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms EXECUTE sofia/external/+14792460547 at flowroute.com playback(silence_stream://2000) 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [last_fax]=[+14792460547-2013-01-10-13-39-26] EXECUTE sofia/external/+14792460547 at flowroute.com set(last_fax=+14792460547-2013-01-10-13-39-26) 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38_request=true) 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38]=[true] 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completed][200] EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38=true) 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel [sofia/external/+14792460547 at flowroute.com] has been answered 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=sendrecv a=ptime:20 a=silenceSupp:off - - - - a=fmtp:101 0-16 a=rtpmap:101 telephone-event/8000 a=rtpmap:0 PCMU/8000 m=audio 27588 RTP/AVP 0 101 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP sofia/external/+14792460547 at flowroute.com: 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success [72.201.112.131]:[27588] EXECUTE sofia/external/+14792460547 at flowroute.com answer() 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=${last_fax} messages='result: ${fax_result_text} sender:${fax_remote_station_id} pages:${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] EXECUTE sofia/external/+14792460547 at flowroute.com set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} messages='result: \${fax_result_text} sender:\${fax_remote_station_id} pages:\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/+14792460547/3150) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [call_direction]=[local] EXECUTE sofia/external/+14792460547 at flowroute.com set(call_direction=local) 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() Dialplan: sofia/external/+14792460547 at flowroute.com Action rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) Dialplan: sofia/external/+14792460547 at flowroute.com Action playback(silence_stream://2000) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38_request=true) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38=true) Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() Dialplan: sofia/external/+14792460547 at flowroute.com Action set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} pages:\\\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\\\${caller_id_name}' caller_id_number=\\\${caller_id_number}) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test Fax] destination_number(3150) =~ /^3150$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Test Fax] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain_user] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->send_to_voicemail] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->extension-intercom] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->intercept-ext] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_return] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_privacy] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [global] Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->global] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [redial] destination_number(3150) =~ /^(redial|\*870)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->redial] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->group-intercept] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action set(call_direction=local) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call-direction] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Flowroute.11d] continue=false 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->3150 in context default 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer sofia/external/+14792460547 at flowroute.com to XML[3150 at default] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML default) 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] EXECUTE sofia/external/+14792460547 at flowroute.com export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [outside_call]=[true] EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 XML default) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] context(public) =~ /public/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->Test_Number] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->call_debug] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(outside_call=true) Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [outside_call] Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->outside_call] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->unloop] continue=false 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->15203039810 in context public 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+14792460547 at flowroute.com) Callstate Change DOWN -> RINGING 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+14792460547 at flowroute.com SOFIA INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf send/recv payload to 101 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 sofia/external/+14792460547 at flowroute.com Original read codec set to PCMU:0 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples 64000 bits 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] a=ptime:20 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=rtpmap:18 G729/8000 m=audio 31632 RTP/AVP 0 18 101 t=0 0 c=IN IP4 66.42.121.5 s=- o=- 275426305 1357849875 IN IP4 66.42.121.5 v=0 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [received][100] 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 (sofia/external/+14792460547 at flowroute.com) State NEW 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel sofia/external/+14792460547 at flowroute.com [d276f012-5b65-11e2-beae-8564ff827a25] Robert Date: Thu, 10 Jan 2013 09:47:18 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Can you pastebin a console debug log w/ sip trace of a call failing to negotiate t.38 correctly? That might help us figure out where to look. -MC On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: Just installed FusionPBX using the script built for a Centos 6 install. Everything works great in regards to calls, but my main goal of this project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have tried many things in media bypass, routing to a fax extension, using an ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me in the right direction as to where I should begin? I am guessing it isn't supposed to work right out of the box. I am using Flowroute as my provider right now and the have confirmed that they utilize T38 in both directions. Thanks! Robert _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/0ba8556c/attachment-0001.html From gassaad at emassembly.com Wed Jan 16 20:11:34 2013 From: gassaad at emassembly.com (George Assaad) Date: Wed, 16 Jan 2013 12:11:34 -0500 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: Message-ID: The reason for outbound extension is: When you send a fax from ATA to the outside world, it will dial first to the outbound extension and then after successful receive it will dial out to the intended number. As for the error you have it may be related to your ATA. I am not sure please see why do you get the following message. "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig wrote: > OK, I've already tried creating one extension for inbound and set an email > address. Of course, it failed and the errors are at the bottom of this > email. What is the purpose of the outbound extension? > > Robert Craig > > Fort Huachuca, AZ 85613 > > 520-226-9505 (Home) > > 760-583-8270 (Cell) > > > ------------------------------ > Date: Wed, 16 Jan 2013 09:57:51 -0500 > From: gassaad at emassembly.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Sorry for the delay to get back to you. > 1- App --> fax server > 2- Create two extensions (one for inbound) another for (outbound) From > within either extension you can send faxes "virtual fax" > 3- point your create your outbound and inbound dialplans as normal but > make it to point to those extensions. > > Good luck, > > George > > On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig wrote: > > What did you do on the config side to get it to work? I've tried both an > Ata and a fax to email extension and they both fail in both directions. > > Robert > > On Jan 12, 2013, at 2:04 PM, "George Assaad" > wrote: > > I use Fusion successfully to send / receive fax. > The way it works: > ATA --->FaxServer. > After successful receipt, Fax Server---> DID where you > have dialled originally. > > Cheers, > > George > > > On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig wrote: > > Did anyone see the issue or what I may have missed in the config? As > mentioned earlier, I thought it would after a fresh install. Any help is > appreciated. > > Robert > > > ------------------------------ > From: craig.robert at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 10 Jan 2013 13:43:09 -0700 > > Subject: Re: [Freeswitch-users] T38 Configuration > > Thanks for the quick response. Below is the capture from the log viewer. > In this test, I am using a remote fax machine (line from Cox with physical > machine) faxing into the "email server" where I created an extension for > fax-to-email. Calling number is 1479-246-0547. > > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 > (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 > sofia/external/+14792460547 at flowroute.com Standard DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/ > +14792460547 at flowroute.com SOFIA DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 > (sofia/external/+14792460547 at flowroute.com) State DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 > (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> > DOWN > 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close > Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] > 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 > (sofia/external/+14792460547 at flowroute.com) Ended > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 > (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external > entities > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> > CS_DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 > (sofia/external/+14792460547 at flowroute.com) State REPORTING going to > sleep > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 > sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: > NORMAL_CLEARING > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 > (sofia/external/+14792460547 at flowroute.com) State REPORTING > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_REPORTING > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 > (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> > CS_REPORTING > > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup > Command with no Session system(/usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 > name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after > permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 > caller_id_name='' caller_id_number=): > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 > (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 > sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: > NORMAL_CLEARING > 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to > sofia/external/+14792460547 at flowroute.com > 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel sofia/external/+ > 14792460547 at flowroute.com hanging up, cause: NORMAL_CLEARING > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 > (sofia/external/+14792460547 at flowroute.com) State HANGUP > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_HANGUP > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 > sofia/external/+14792460547 at flowroute.com skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal > sofia/external/+14792460547 at flowroute.com [KILL] > 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup > sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] > 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+ > 14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP > EXECUTE sofia/external/+14792460547 at flowroute.com hangup() > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 sofia/external/+ > 14792460547 at flowroute.com Restore previous codec PCMU:0. > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 > ============================================================================== > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: > 14400 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image resolution: > 0x0 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: > 0 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages > transferred: 0 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station id: > SpanDSP Fax Ident > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station > id: ATW-4792460547 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing > not successful - result (48) Disconnected after permitted retries. > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 > ============================================================================== > 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port > confirmed. > 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+ > 14792460547 at flowroute.com image media sdp: > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+ > 14792460547 at flowroute.com image media sdp: > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+ > 14792460547 at flowroute.com Set rtp dtmf delay to 40 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive > payload to 101 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send > payload to 101 > 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer > spec [0] must be between 20 and 10000 > 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] > 160 bytes per 20ms > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP > [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> > 66.42.121.5 port 31632 codec: 0 ms: 20 > 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [ready][200] > 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxRateManagement:transferredTCF > m=image 31632 udptl t38 > t=0 0 > c=IN IP4 66.42.121.5 > s=- > o=- 275426305 1357849876 IN IP4 66.42.121.5 > v=0 > 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: > 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [completing][200] > 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+ > 14792460547 at flowroute.com Update Callee ID to "+14792460547" <+14792460547 > > > 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [calling][0] > 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+ > 14792460547 at flowroute.com image media sdp: > 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec > activation Success L16 > 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 sofia/external/+ > 14792460547 at flowroute.com Push codec L16:70 > 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec > activation Success L16 20000 > EXECUTE sofia/external/+14792460547 at flowroute.comrxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done playing > file silence_stream://2000 > 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port > confirmed. > 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [ready][200] > 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec > Activated L16 at 8000hz 1 channels 20ms > EXECUTE sofia/external/+14792460547 at flowroute.complayback(silence_stream://2000) > 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET > [last_fax]=[+14792460547-2013-01-10-13-39-26] > EXECUTE sofia/external/+14792460547 at flowroute.comset(last_fax=+14792460547-2013-01-10-13-39-26) > 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] > EXECUTE sofia/external/+14792460547 at flowroute.comset(fax_enable_t38_request=true) > 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [fax_enable_t38]=[true] > 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [completed][200] > EXECUTE sofia/external/+14792460547 at flowroute.comset(fax_enable_t38=true) > 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel > [sofia/external/+14792460547 at flowroute.com] has been answered > 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+ > 14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE > 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=sendrecv > a=ptime:20 > a=silenceSupp:off - - - - > a=fmtp:101 0-16 > a=rtpmap:101 telephone-event/8000 > a=rtpmap:0 PCMU/8000 > m=audio 27588 RTP/AVP 0 101 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP > sofia/external/+14792460547 at flowroute.com: > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+ > 14792460547 at flowroute.com Set rtp dtmf delay to 40 > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive > payload to 101 > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send > payload to 101 > 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] > 160 bytes per 20ms > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP > [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> > 66.42.121.5 port 31632 codec: 0 ms: 20 > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success > [72.201.112.131]:[27588] > EXECUTE sofia/external/+14792460547 at flowroute.com answer() > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 name=${last_fax} > messages='result: ${fax_result_text} sender:${fax_remote_station_id} > pages:${fax_document_total_pages}' domain=192.168.4.15 > caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] > EXECUTE sofia/external/+14792460547 at flowroute.comset(api_hangup_hook=system /usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} > messages='result: \${fax_result_text} sender:\${fax_remote_station_id} > pages:\${fax_document_total_pages}' domain=192.168.4.15 > caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 > -0700] > EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, > 10 Jan 2013 13:39:25 -0700) > EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-last_dial/+14792460547/3150) > EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-spymap/ > +14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [call_direction]=[local] > EXECUTE sofia/external/+14792460547 at flowroute.comset(call_direction=local) > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 > sofia/external/+14792460547 at flowroute.com Standard EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+ > 14792460547 at flowroute.com SOFIA EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 > (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> > CS_EXECUTE > Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() > Dialplan: sofia/external/+14792460547 at flowroute.com Action > rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > playback(silence_stream://2000) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(fax_enable_t38_request=true) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(fax_enable_t38=true) > Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(api_hangup_hook=system /usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} > messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} > pages:\\\${fax_document_total_pages}' domain=192.168.4.15 > caller_id_name='\\\${caller_id_name}' > caller_id_number=\\\${caller_id_number}) > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test > Fax] destination_number(3150) =~ /^3150$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->Test Fax] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->vmain_user] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] > destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->vmain] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->send_to_voicemail] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->extension-intercom] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->intercept-ext] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->call_return] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->call_privacy] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition > [global] > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->global] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [redial] > destination_number(3150) =~ /^(redial|\*870)$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->redial] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->group-intercept] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action > set(call_direction=local) > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->call-direction] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->Flowroute.11d] continue=false > 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing > +14792460547 <+14792460547>->3150 in context default > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 > sofia/external/+14792460547 at flowroute.com Standard ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+ > 14792460547 at flowroute.com SOFIA ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep > 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer > sofia/external/+14792460547 at flowroute.com to XML[3150 at default] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+ > 14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING > EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML > default) > 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT > (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > EXECUTE sofia/external/+14792460547 at flowroute.comexport(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [outside_call]=[true] > EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 > sofia/external/+14792460547 at flowroute.com Standard EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+ > 14792460547 at flowroute.com SOFIA EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 > (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> > CS_EXECUTE > Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 > XML default) > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (PASS) [Test_Number] context(public) =~ /public/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->Test_Number] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->call_debug] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(outside_call=true) > Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition > [outside_call] > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->outside_call] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->unloop] continue=false > 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing > +14792460547 <+14792460547>->15203039810 in context public > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 > sofia/external/+14792460547 at flowroute.com Standard ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+ > 14792460547 at flowroute.com SOFIA ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING > 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+ > 14792460547 at flowroute.com) Callstate Change DOWN -> RINGING > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_ROUTING > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 > (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+ > 14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING > 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+ > 14792460547 at flowroute.com SOFIA INIT > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 > (sofia/external/+14792460547 at flowroute.com) State INIT > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+ > 14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf > send/recv payload to 101 > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 sofia/external/+ > 14792460547 at flowroute.com Original read codec set to PCMU:0 > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec > sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples > 64000 bits > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > > a=ptime:20 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=fmtp:18 annexb=no > a=rtpmap:18 G729/8000 > m=audio 31632 RTP/AVP 0 18 101 > t=0 0 > c=IN IP4 66.42.121.5 > s=- > o=- 275426305 1357849875 IN IP4 66.42.121.5 > v=0 > 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: > 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [received][100] > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 > (sofia/external/+14792460547 at flowroute.com) State NEW > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel > sofia/external/+14792460547 at flowroute.com[d276f012-5b65-11e2-beae-8564ff827a25] > > Robert > > > ------------------------------ > Date: Thu, 10 Jan 2013 09:47:18 -0800 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Can you pastebin a console debug log w/ sip trace of a call failing to > negotiate t.38 correctly? That might help us figure out where to look. > -MC > > On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: > > Just installed FusionPBX using the script built for a Centos 6 install. > Everything works great in regards to calls, but my main goal of this > project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have > tried many things in media bypass, routing to a fax extension, using an > ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me > in the right direction as to where I should begin? I am guessing it isn't > supposed to work right out of the box. I am using Flowroute as my provider > right now and the have confirmed that they utilize T38 in both directions. > Thanks! > > > Robert > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/827e2a9f/attachment-0001.html From krice at freeswitch.org Wed Jan 16 20:30:31 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 16 Jan 2013 11:30:31 -0600 Subject: [Freeswitch-users] compile error In-Reply-To: <20130116085100.259790@gmx.com> Message-ID: Jira please.... On 1/16/13 2:51 AM, "peter ness" wrote: > Can anybody advise why I am getting this error in compiling. This is a fresh > download fron git and default modules.conf and ./configure Systen is amd > x86_64-pc-linux-gnu make[2]: Leaving directory > `/root/Desktop/freeswitch/libs/curl' make[1]: Leaving directory > `/root/Desktop/freeswitch/libs/curl' which: no git in > (/usr/local/sbin:/usr/sbin:/sbin:/usr/local/bin:/usr/bin:/bin:/usr/games:/usr/ > lib64/kde4/libexec:/usr/lib64/qt/bin) cat > /root/Desktop/freeswitch/src/include/switch_cpp.h | perl > /root/Desktop/freeswitch/build/strip.pl > > /root/Desktop/freeswitch/src/include/switch_swigable_cpp.h make > "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" > /root/Desktop/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; > else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; > echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v > "#" /root/Desktop/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq > )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; > done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then > tmp_mods="$(grep -v "#" /root/Desktop/freeswitch/modules.conf | sed -e > "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in > $tmp_mods ; do echo $i-install ; done)"; echo $mods )" > "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" > /root/Desktop/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; > else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; > done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" > /root/Desktop/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | > sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo > $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" > /root/Desktop/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | > sort | uniq )"; ?mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo > $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" > /root/Desktop/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | > sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo > $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" > /root/Desktop/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | > sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; > echo $mods )" `test -n "" || echo -s` all-recursive quiet_libtool: compile: > ?gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c > -I/root/Desktop/freeswitch/libs/curl/include > -I/root/Desktop/freeswitch/src/include -I/root/Desktop/freeswitch/src/include > -I/root/Desktop/freeswitch/libs/libteletone/src > -I/root/Desktop/freeswitch/libs/stfu -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > -Ilibs/sofia-sip/libsofia-sip-ua/sdp/sofia-sip > -Ilibs/sofia-sip/libsofia-sip-ua/su -g -O2 -pthread -D_REENTRANT -D_GNU_SOURCE > -I/root/Desktop/freeswitch/libs/apr/include > -I/root/Desktop/freeswitch/libs/apr-util/include > -I/root/Desktop/freeswitch/libs/apr-util/xml/expat/lib > -I/root/Desktop/freeswitch/libs/libtpl-1.5/src > -I/root/Desktop/freeswitch/libs/stfu -I/root/Desktop/freeswitch/libs/sqlite > -I/root/Desktop/freeswitch/libs/pcre > -I/root/Desktop/freeswitch/libs/speex/include -Ilibs/speex/include > -I/root/Desktop/freeswitch/libs/srtp/include > -I/root/Desktop/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include > -I/root/Desktop/freeswitch/libs/spandsp/src > -I/root/Desktop/freeswitch/libs/tiff-4.0.2/libtiff -DENABLE_SRTP > -DSWITCH_HAVE_ODBC -I/usr/local/include > -I/root/Desktop/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > -I/root/Desktop/freeswitch/libs/curl/include > -I/root/Desktop/freeswitch/src/include -I/root/Desktop/freeswitch/src/include > -I/root/Desktop/freeswitch/libs/libteletone/src > -I/root/Desktop/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall > -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT > libfreeswitch_la-switch_apr.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_apr.Tpo -c src/switch_apr.c ?-fPIC -DPIC -o > .libs/libfreeswitch_la-switch_apr.o In file included from > /root/Desktop/freeswitch/libs/spandsp/src/spandsp.h:66:0, ? ? ? ? ? ? ? ? > from ./src/include/private/switch_core_pvt.h:35, ? ? ? ? ? ? ? ? from > src/switch_apr.c:37: /root/Desktop/freeswitch/libs/spandsp/src/spandsp/fir.h: > In function > 'fir16_create': /root/Desktop/freeswitch/libs/spandsp/src/spandsp/fir.h:94:5: > error: implicit declaration of function 'rpl_malloc' > [-Werror=implicit-function-declaration] /root/Desktop/freeswitch/libs/spandsp/ > src/spandsp/fir.h:94:25: error: cast to pointer from integer of different size > [-Werror=int-to-pointer-cast] /root/Desktop/freeswitch/libs/spandsp/src/spands > p/fir.h: In function > 'fir32_create': /root/Desktop/freeswitch/libs/spandsp/src/spandsp/fir.h:213:20 > : error: cast to pointer from integer of different size > [-Werror=int-to-pointer-cast] /root/Desktop/freeswitch/libs/spandsp/src/spands > p/fir.h: In function > 'fir_float_create': /root/Desktop/freeswitch/libs/spandsp/src/spandsp/fir.h:26 > 1:20: error: cast to pointer from integer of different size > [-Werror=int-to-pointer-cast] src/switch_apr.c: In function > 'switch_dir_open': src/switch_apr.c:558:22: error: initialization makes > pointer from integer without a cast [-Werror] cc1: all warnings being treated > as errors make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make: *** > [all] Error > 2 _________________________________________________________________________ P > rofessional FreeSWITCH Consulting > Services: consulting at freeswitch.org http://www.freeswitchsolutions.com FreeSW > ITCH-powered IP PBX: The CudaTel Communication > Server Official FreeSWITCH > Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon. > com FreeSWITCH-users mailing > list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From craig.robert at hotmail.com Wed Jan 16 20:40:31 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Wed, 16 Jan 2013 10:40:31 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: , , , , , , , , Message-ID: OK, I am not extremely concerned with getting an ATA working right now. If I could just get an inbound extension-to-email working it would be great. I'll start from scratch and then let you know what happens. Just to make sure, I am going to create an extension, assign an email address to it, make sure there is an inbound route to that extension, and then test it. Robert Date: Wed, 16 Jan 2013 12:11:34 -0500 From: gassaad at emassembly.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration The reason for outbound extension is:When you send a fax from ATA to the outside world, it will dial first to the outbound extension and then after successful receive it will dial out to the intended number. As for the error you have it may be related to your ATA. I am not sure please see why do you get the following message."Invalid Jitterbuffer spec [0] must be between 20 and 10000 " On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig wrote: OK, I've already tried creating one extension for inbound and set an email address. Of course, it failed and the errors are at the bottom of this email. What is the purpose of the outbound extension? Robert Date: Wed, 16 Jan 2013 09:57:51 -0500 From: gassaad at emassembly.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Sorry for the delay to get back to you. 1- App --> fax server2- Create two extensions (one for inbound) another for (outbound) From within either extension you can send faxes "virtual fax" 3- point your create your outbound and inbound dialplans as normal but make it to point to those extensions. Good luck, George On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig wrote: What did you do on the config side to get it to work? I've tried both an Ata and a fax to email extension and they both fail in both directions. Robert On Jan 12, 2013, at 2:04 PM, "George Assaad" wrote: I use Fusion successfully to send / receive fax.The way it works: ATA --->FaxServer.After successful receipt, Fax Server---> DID where you have dialled originally. Cheers, George On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig wrote: Did anyone see the issue or what I may have missed in the config? As mentioned earlier, I thought it would after a fresh install. Any help is appreciated. Robert From: craig.robert at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Thu, 10 Jan 2013 13:43:09 -0700 Subject: Re: [Freeswitch-users] T38 Configuration Thanks for the quick response. Below is the capture from the log viewer. In this test, I am using a remote fax machine (line from Cox with physical machine) faxing into the "email server" where I created an extension for fax-to-email. Calling number is 1479-246-0547. 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 sofia/external/+14792460547 at flowroute.com Standard DESTROY 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/+14792460547 at flowroute.com SOFIA DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> DOWN 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 (sofia/external/+14792460547 at flowroute.com) Ended 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external entities 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> CS_DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING going to sleep 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: NORMAL_CLEARING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> CS_REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup Command with no Session system(/usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 caller_id_name='' caller_id_number=): 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: NORMAL_CLEARING 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to sofia/external/+14792460547 at flowroute.com 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel sofia/external/+14792460547 at flowroute.com hanging up, cause: NORMAL_CLEARING 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_HANGUP 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 sofia/external/+14792460547 at flowroute.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal sofia/external/+14792460547 at flowroute.com [KILL] 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP EXECUTE sofia/external/+14792460547 at flowroute.com hangup() 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 14400 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station id: ATW-4792460547 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (48) Disconnected after permitted retries. 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer spec [0] must be between 20 and 10000 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxRateManagement:transferredTCF m=image 31632 udptl t38 t=0 0 c=IN IP4 66.42.121.5 s=- o=- 275426305 1357849876 IN IP4 66.42.121.5 v=0 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completing][200] 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+14792460547 at flowroute.com Update Callee ID to "+14792460547" <+14792460547> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [calling][0] 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec activation Success L16 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 sofia/external/+14792460547 at flowroute.com Push codec L16:70 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec activation Success L16 20000 EXECUTE sofia/external/+14792460547 at flowroute.com rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done playing file silence_stream://2000 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms EXECUTE sofia/external/+14792460547 at flowroute.com playback(silence_stream://2000) 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [last_fax]=[+14792460547-2013-01-10-13-39-26] EXECUTE sofia/external/+14792460547 at flowroute.com set(last_fax=+14792460547-2013-01-10-13-39-26) 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38_request=true) 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38]=[true] 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completed][200] EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38=true) 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel [sofia/external/+14792460547 at flowroute.com] has been answered 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=sendrecv a=ptime:20 a=silenceSupp:off - - - - a=fmtp:101 0-16 a=rtpmap:101 telephone-event/8000 a=rtpmap:0 PCMU/8000 m=audio 27588 RTP/AVP 0 101 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP sofia/external/+14792460547 at flowroute.com: 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success [72.201.112.131]:[27588] EXECUTE sofia/external/+14792460547 at flowroute.com answer() 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=${last_fax} messages='result: ${fax_result_text} sender:${fax_remote_station_id} pages:${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] EXECUTE sofia/external/+14792460547 at flowroute.com set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} messages='result: \${fax_result_text} sender:\${fax_remote_station_id} pages:\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/+14792460547/3150) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [call_direction]=[local] EXECUTE sofia/external/+14792460547 at flowroute.com set(call_direction=local) 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() Dialplan: sofia/external/+14792460547 at flowroute.com Action rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) Dialplan: sofia/external/+14792460547 at flowroute.com Action playback(silence_stream://2000) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38_request=true) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38=true) Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() Dialplan: sofia/external/+14792460547 at flowroute.com Action set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} pages:\\\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\\\${caller_id_name}' caller_id_number=\\\${caller_id_number}) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test Fax] destination_number(3150) =~ /^3150$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Test Fax] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain_user] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->send_to_voicemail] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->extension-intercom] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->intercept-ext] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_return] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_privacy] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [global] Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->global] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [redial] destination_number(3150) =~ /^(redial|\*870)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->redial] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->group-intercept] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action set(call_direction=local) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call-direction] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Flowroute.11d] continue=false 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->3150 in context default 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer sofia/external/+14792460547 at flowroute.com to XML[3150 at default] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML default) 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] EXECUTE sofia/external/+14792460547 at flowroute.com export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [outside_call]=[true] EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 XML default) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] context(public) =~ /public/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->Test_Number] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->call_debug] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(outside_call=true) Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [outside_call] Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->outside_call] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->unloop] continue=false 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->15203039810 in context public 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+14792460547 at flowroute.com) Callstate Change DOWN -> RINGING 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+14792460547 at flowroute.com SOFIA INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf send/recv payload to 101 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 sofia/external/+14792460547 at flowroute.com Original read codec set to PCMU:0 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples 64000 bits 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] a=ptime:20 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=rtpmap:18 G729/8000 m=audio 31632 RTP/AVP 0 18 101 t=0 0 c=IN IP4 66.42.121.5 s=- o=- 275426305 1357849875 IN IP4 66.42.121.5 v=0 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [received][100] 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 (sofia/external/+14792460547 at flowroute.com) State NEW 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel sofia/external/+14792460547 at flowroute.com [d276f012-5b65-11e2-beae-8564ff827a25] Robert Date: Thu, 10 Jan 2013 09:47:18 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Can you pastebin a console debug log w/ sip trace of a call failing to negotiate t.38 correctly? That might help us figure out where to look. -MC On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: Just installed FusionPBX using the script built for a Centos 6 install. Everything works great in regards to calls, but my main goal of this project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have tried many things in media bypass, routing to a fax extension, using an ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me in the right direction as to where I should begin? I am guessing it isn't supposed to work right out of the box. I am using Flowroute as my provider right now and the have confirmed that they utilize T38 in both directions. Thanks! Robert _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/455b4c55/attachment-0001.html From msc at freeswitch.org Wed Jan 16 20:45:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Jan 2013 09:45:37 -0800 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today Message-ID: Hello all! Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_16 We have updates on FreeSWITCH 1.2.6, the new FreeSWITCH book, and ClueCon 2012 videos. We are also going to discuss some techniques for how to validate bugs and update Jira tickets with the information. Also, if anyone is familiar with git bisect we would like to invite you to spend a few minutes giving us some pointers as that is a very useful tool for locating the commit that "broke something." Talk to you shortly! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/326862ff/attachment.html From steveayre at gmail.com Wed Jan 16 21:14:21 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 16 Jan 2013 18:14:21 +0000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> Message-ID: > > However under freeswitch if we don't start_dtmf before the bridge the > backend PBX boxes don't recognise > the DTMF inband (even though the tones are audible ie you can hear them on > a call recording on the > PBX). > > Have we missed something here? We would have thought with inband DTMF on > non compressed codec (no > transcoding) that the tones would just work with the media stream? start_dtmf will detect inband DTMF, and then send out of band on the outgoing leg of the bridge. This is expected. You'll will need to call start_dtmf from dialplan, which can easily be done from dialplan by checking for some condition identifying calls requiring it. If they're authenticating to FS then it'd be trivial to set a variable in their user directory entry that you can test in the dialplan. Now this is just a guess... During the codec negotiation FS will tell your PBX that it supports telephone-event (RFC2833). It may be that because the PBX sees that in the SDP that it doesn't look for inband DTMF, while when it received the SDP direct from the customer it didn't contain telephone-event and so did check for inband DTMF. Indeed it'd probably be a good idea for them to do so, since if you receive the same DTMF digit both inband and through RFC2833 then you'd be duplicating digits. So my guess is that the PBX doesn't bother check for inband because FS tells it it's sending out-of-band, which'd differ from before. -Steve -Steve On 16 January 2013 04:02, support at ecn.net.au wrote: > Hi All > > > > We're quite new to Freeswitch and are in the process of migrating from > OpenSer (as an SBC) to Freeswitch. > > > > Mostly all is working well, except an oddity on DTMF. > > > > Our scenario: > > > > Telco/SIP Provider A is passing us calls using DTMF inband. > > > > We have a freeswitch configured as a SBC using 2 sip profiles (telco and > internal) to topology hide and manage > > distribution of calls to the PBX servers located behind the SBC. > > > > The freeswitch will be handling up to a few hundred calls so we're trying to > keep it lightweight. > > > > Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling > customer needs. > > > > An example inbound call profile looks like this: > > > > > > > > > > > > data="sofia/external/123456 at INTERNAL.PBX.IP:5060"/> > > > > > > > > Initially when calling into the platform IVR type applications runinng on > our PBX boxes would not > > work (you could hear the DTMF but the platform did not recognise the tones). > > > > We have had to add the appliation start_dtmf in order for Freeswitch to pass > the DTMF to the Asterisk > > PBX behind the SBC. Interestingly on our OpenSer platform we just proxied > the media (rtpproxy) with > > inband DTMF from the Telco and our PBX boxes recognised the inband DTMF > tones on the PBX platforms and > > IVR type applications just worked. > > > > However under freeswitch if we don't start_dtmf before the bridge the > backend PBX boxes don't recognise > > the DTMF inband (even though the tones are audible ie you can hear them on a > call recording on the > > PBX). > > > > Have we missed something here? We would have thought with inband DTMF on > non compressed codec (no > > transcoding) that the tones would just work with the media stream? > > > > We have confirmed both legs are PCMA and when using start_dtmf the first > second of the call is clipped. > > > > > > > > Kind Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/190ffa65/attachment.html From steveayre at gmail.com Wed Jan 16 21:17:50 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 16 Jan 2013 18:17:50 +0000 Subject: [Freeswitch-users] Status of mod_lua and dbh.release() In-Reply-To: References: Message-ID: > > We're seeing quite a free PostgreSQL 9.1 processes left. How many is 'quite a few' (I assume that's what you meant). The DB pool maintains a number of option persistent connections that are reused - it would be normal to see a number of idle connections still open. On 15 January 2013 21:03, Gavin Henry wrote: > Hi all, > > Does this release the db handle like it says? > > dbh:release() (optional) releases the handle back to the pool so it > can be re-used by another thread. This is also automatically done when > the dbh goes out of scope and is garbage collected (for example when > your script returns). > > We're seeing quite a free PostgreSQL 9.1 processes left. > > -- > http://www.suretecsystems.com/services/openldap/ > http://www.surevoip.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/b000a8bf/attachment.html From jaybinks at gmail.com Thu Jan 17 01:04:43 2013 From: jaybinks at gmail.com (jay binks) Date: Thu, 17 Jan 2013 08:04:43 +1000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <8A80C99638A5E24E83696EAE33A9B6B968B53027EF@twbexmbx001.ecn.hosting> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027EF@twbexmbx001.ecn.hosting> Message-ID: Id be suggesting that you get PCaps on all legs of the call and just be 100% sure where the audio is clipped. On 16 January 2013 22:55, support at ecn.net.au wrote: > Hi > > This is what we now have, however there is an interesting side effect > (major issues actally) we're getting when using start_dtmf on the aleg from > the "dodgy" telco. the oddity is that when start_dtmf is executed prior to > bridge some audio content from the back end PBX's (behind the Freeswitch > SBC) does not seem to transmit back through to the caller (on the aleg of > the freeswitch SBC (for example play back of a canned audio file from an > IVR on an Asterisk PBX behind the SBC). > > In testing Freeswitch on version 1.2 (1.2.5 and 1.2.3) and 1.3 testing all > versions have this issue, however on an old 1.0.6 legacy version that we > compiled today - does not produce the same problem (it works perfectly); > could it be a bug in the 1.2/3 freeswitch do you think? Or is it something > we've missed? > > To reconfirm: > > context A (telco context) we have a sip profile (dtmf_mode=none), > context B (to.pbx context) we have a sip profile (dtmf_mode=rfc2833) > > On the xml dial plan for context A we start_dtmf before we bridge the call > from the telco to the backend pbx. > > When executing start_dtmf the system correctly transmits the dtmf digits > to the backend pbx's however the by product (which we earlier thought was > clipping the start of the call) is that some audio content (such as some > audio file playback) doesn't get heard by the caller (dialed via the telco > through the SBC to the pbx). > > Any help appreciated! Is there a change in how start_dtmf effects calls > from the 1.2 and onward versions? > > Regards > Mark > > > > > ________________________________________ > From: jay binks [jaybinks at gmail.com] > Sent: Wednesday, 16 January 2013 7:17 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SBC In-band DTMF > > on your internal sip profile ( one facing PBX's ) set name="dtmf-type" value="rfc2833"/> > > then note the dialplan context that super dodgey sip carrier ( the one > doing inband only ) > sends calls to, and use on calls from > them. > > Id advise against in that location > unless you REALLY want that. > > that should do what you want. > > > On 16 January 2013 14:02, support at ecn.net.au < > support at ecn.net.au> wrote: > > Hi All > > > > We're quite new to Freeswitch and are in the process of migrating from > OpenSer (as an SBC) to Freeswitch. > > > > Mostly all is working well, except an oddity on DTMF. > > > > Our scenario: > > > > Telco/SIP Provider A is passing us calls using DTMF inband. > > > > We have a freeswitch configured as a SBC using 2 sip profiles (telco and > internal) to topology hide and manage > > distribution of calls to the PBX servers located behind the SBC. > > > > The freeswitch will be handling up to a few hundred calls so we're trying > to keep it lightweight. > > > > Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling > customer needs. > > > > An example inbound call profile looks like this: > > > > > > > > > > > > > > > > > > > > > Initially when calling into the platform IVR type applications runinng on > our PBX boxes would not > > work (you could hear the DTMF but the platform did not recognise the > tones). > > > > We have had to add the appliation start_dtmf in order for Freeswitch to > pass the DTMF to the Asterisk > > PBX behind the SBC. Interestingly on our OpenSer platform we just > proxied the media (rtpproxy) with > > inband DTMF from the Telco and our PBX boxes recognised the inband DTMF > tones on the PBX platforms and > > IVR type applications just worked. > > > > However under freeswitch if we don't start_dtmf before the bridge the > backend PBX boxes don't recognise > > the DTMF inband (even though the tones are audible ie you can hear them on > a call recording on the > > PBX). > > > > Have we missed something here? We would have thought with inband DTMF on > non compressed codec (no > > transcoding) that the tones would just work with the media stream? > > > > We have confirmed both legs are PCMA and when using start_dtmf the first > second of the call is clipped. > > > > > Kind Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/ddf51845/attachment-0001.html From schoch+freeswitch.org at xwin32.com Thu Jan 17 02:24:59 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 16 Jan 2013 15:24:59 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: <20130116025242.83fde4e4@mail.tritonwest.net> References: <20130116025242.83fde4e4@mail.tritonwest.net> Message-ID: On Tue, Jan 15, 2013 at 6:52 PM, Dave R. Kompel wrote: > ** > Another option would be on the Linux box, enable SPANDSP modems, and use > com0com on windows to backup com ports to raw TCP connections. It supports > both. That way it would talk over a TCP pipe to an emulated modem on your > other FreeSWITCH box. > That sounds like the way to go. I'll give that a try. But first, I have another FAX problem. The users here use a real FAX machine to send FAXes. Since we're getting rid of the POTS lines, I attempted to use a (cheap) OBi100 to connect the analog FAX machine to the FreeSwitch box. It didn't work. Voice calls work, but it can't seem to sync with the remote FAX machine. Any ideas how to get a regular FAX machine to work with VoIP? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/1e664ccd/attachment.html From paan.singh.coder at gmail.com Wed Jan 16 22:20:52 2013 From: paan.singh.coder at gmail.com (Paan Singh) Date: Thu, 17 Jan 2013 00:50:52 +0530 Subject: [Freeswitch-users] 488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH Message-ID: Hi, I'm using JsSIP from a webpage to make a SIP call to FS, using OverSIP as a Websocket->SIP proxy. I'm able to Register successfully, but when I make a call from JsSIP UA to FreeSWITCH, I get a 180 Ringing, but after that I get a 488 Not Acceptable here.. Please help me figure out what I'm doing wrong here. My inbound_codec_prefs is : Here is FreeSWITCH's log for the incoming call : ============================================= port.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg 0x2913be0 from (udp/116.214.15.45:5080) has 3113 bytes, veclen = 1 tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x2913be0 (3113 bytes) from udp/121.245.35.150:5080/sip next=(nil) nta.c:2803 agent_recv_request() nta: received INVITE sip:30201009 at 116.214.15.45 SIP/2.0 (CSeq 2141) nta.c:3161 agent_aliases() nta: canonizing sip:30201009 at 116.214.15.45 with contact nta.c:3002 agent_recv_request() nta: INVITE (2141) going to a default leg nta.c:1348 set_timeout() nta: timer set to 200 ms nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_stack.c:899 nh_create() nua: nh_create: entering nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:282 soa_clone() soa_clone(static::0x26122e0, 0x2606e20, 0x7f91840c4c50) called soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) called nta.c:4313 nta_leg_tcreate() nta_leg_tcreate(0x7f91840356d0) soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f91840b25e0) called soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7f91840b25e0, (nil), 0x285676c, 2333) called nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f91840c4c50): adding session usage tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ 121.245.35.150:5065 tport.c:4026 tport_resolve() tport_resolve addrinfo = 121.245.35.150:5065 tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not found by name UDP/121.245.35.150:5065 tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 564 bytes of 564 to udp/ 121.245.35.150:5065 tport.c:3472 tport_send_msg() tport_vsend returned 564 nta.c:6678 incoming_reply() nta: sent 100 Trying for INVITE (2141) nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_invite 100 Trying nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7f91840b25e0, [0x7f91aa98a5e8], [0x7f91aa98a5e0], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2013-01-17 00:25:27.733506 [NOTICE] switch_channel.c:968 New Channel sofia/external/1000 at 116.214.15.45:5080[4aec7fd8-600e-11e2-8706-1365ba01ad52] 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send signal sofia/external/1000 at 116.214.15.45:5080 [BREAK] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send signal sofia/external/1000 at 116.214.15.45:5080 [BREAK] 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:415 (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_NEW 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:433 (sofia/external/1000 at 116.214.15.45:5080) State NEW nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5576 Channel sofia/external/ 1000 at 116.214.15.45:5080 entering state [received][100] 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5587 Remote SDP: v=0 o=- 2957375511 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video m=audio 40111 RTP/SAVPF 103 104 0 8 106 105 13 126 c=IN IP4 121.245.35.150 a=rtcp:40111 IN IP4 121.245.35.150 a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host generation 0 a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host generation 0 a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx generation 0 a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx generation 0 a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host generation 0 a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host generation 0 a=ice-ufrag:qiKF0ErbxaXldKI0 a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw a=ice-options:google-ice a=mid:audio a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:4278252985 cname:fMHr4uVHPcpHFfGu a=ssrc:4278252985 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx a=ssrc:4278252985 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx00 m=video 40111 RTP/SAVPF 100 101 102 c=IN IP4 121.245.35.150 a=rtcp:40111 IN IP4 121.245.35.150 a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host generation 0 a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host generation 0 a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx generation 0 a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx generation 0 a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host generation 0 a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host generation 0 a=ice-ufrag:qiKF0ErbxaXldKI0 a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw a=ice-options:google-ice a=mid:video a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 a=rtpmap:100 VP8/90000 a=rtpmap:101 red/90000 a=rtpmap:102 ulpfec/90000 a=ssrc:169275022 cname:fMHr4uVHPcpHFfGu a=ssrc:169275022 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx a=ssrc:169275022 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx10 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 AES_CM_128_HMAC_SHA1_80 inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2] 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:3155 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:i8vqmBeU4+/sjJBndCbEQ4G87gmpYeVnqxh8Xuzu] 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:5277 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:2994 (sofia/external/1000 at 116.214.15:45:5080) Callstate Change DOWN -> HANGUP 2013-01-17 00:25:27.753511 [NOTICE] sofia.c:5878 Hangup sofia/external/ 1000 at 116.214.15.45:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:3017 Send signal sofia/external/1000 at 116.214.15.45:5080 [KILL] 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal sofia/external/1000 at 116.214.15.45:5080 [BREAK] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_HANGUP 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 (sofia/external/1000 at 116.214.15.45:5080) State HANGUP 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:503 Channel sofia/external/ 1000 at 116.214.15.45:5080 hanging up, cause: INCOMPATIBLE_DESTINATION 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 488 nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_respond 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:48 sofia/external/1000 at 116.214.15.45:5080 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 (sofia/external/1000 at 116.214.15.45:5080) State HANGUP going to sleep nua_stack.c:573 nua_stack_signal() nua(0x7f91840c4c50): recv signal r_respond 488 Not Acceptable Here 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:446 (sofia/external/1000 at 116.214.15.45:5080) State Change CS_HANGUP -> CS_REPORTING nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal sofia/external/1000 at 116.214.15.45:5080 [BREAK] soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) called 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_REPORTING nua_session.c:2316 nua_invite_server_respond() nua: nua_invite_server_respond: entering soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0x7f91840b25e0) called tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ 121.245.35.150:5065 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 (sofia/external/1000 at 116.214.15.45:5080) State REPORTING tport.c:4026 tport_resolve() tport_resolve addrinfo = 121.245.35.150:5065 tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not found by name UDP/121.245.35.150:5065 tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 784 bytes of 784 to udp/ 121.245.35.150:5065 tport.c:3472 tport_send_msg() tport_vsend returned 784 nta.c:6678 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE (2141) nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f91840c4c50): removing session usage nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call state changed: received -> terminated nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state 488 Not Acceptable Here nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_terminated 488 Not Acceptable Here soa.c:358 soa_destroy() soa_destroy(static::0x7f91840b25e0) called nta.c:4366 nta_leg_destroy() nta_leg_destroy(0x7f91840356d0) nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering nua_stack.c:569 nua_stack_signal() nua(0x7f91840c4c50): recv signal r_destroy nta.c:4366 nta_leg_destroy() nta_leg_destroy((nil)) nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_destroy 2013-01-17 00:25:27.753511 [DEBUG] mod_cdr_sqlite.c:102 Writing SQL to DB: INSERT INTO cdr VALUES ("1000","1000","30201009","public","2013-01-17 00:25:27","","2013-01-17 00:25:27",0,0,"INCOMPATIBLE_DESTINATION","4aec7fd8-600e-11e2-8706-1365ba01ad52","","") 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:92 sofia/external/1000 at 116.214.15.45:5080 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 (sofia/external/1000 at 116.214.15.45:5080) State REPORTING going to sleep 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:440 (sofia/external/1000 at 116.214.15.45:5080) State Change CS_REPORTING -> CS_DESTROY 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal sofia/external/1000 at 116.214.15.45:5080 [BREAK] 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1499 Session 1794 (sofia/external/1000 at 116.214.15.45:5080) Locked, Waiting on external entities 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1517 Session 1794 (sofia/external/1000 at 116.214.15.45:5080) Ended 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1521 Close Channel sofia/external/1000 at 116.214.15.45:5080 [CS_DESTROY] 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:556 (sofia/external/1000 at 116.214.15.45:5080) Callstate Change HANGUP -> DOWN 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:559 (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_DESTROY 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 (sofia/external/1000 at 116.214.15.45:5080) State DESTROY 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:396 sofia/external/ 1000 at 116.214.15.45:5080 SOFIA DESTROY 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:99 sofia/external/1000 at 116.214.15.45:5080 Standard DESTROY 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 (sofia/external/1000 at 116.214.15.45:5080) State DESTROY going to sleep nta.c:1294 agent_timer() nta: timer set next to 322 ms tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg 0x7f9198220070 from (udp/116.214.15.45:5080) has 339 bytes, veclen = 1 tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x7f9198220070 (339 bytes) from udp/121.245.35.150:5080/sip next=(nil) nta.c:2803 agent_recv_request() nta: received ACK sip:30201009 at 116.214.15.45SIP/2.0 (CSeq 2141) nta.c:2938 agent_recv_request() nta: ACK (2141) is going to INVITE (2141) nta.c:1294 agent_timer() nta: timer set next to 4823 ms nta.c:7010 _nta_incoming_timer() nta: timer I fired, terminate 488 response nta.c:5718 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7f91aa98ac20) nta.c:7061 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta.c:1279 agent_timer() nta: timer not set ================================================================ Here is the Log from JsSIP in the browser that receives messages from FreeSWITCH : =========================================================== JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 180 Ringing Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 Record-Route: Record-Route: From: < sip:1000 at 116.214.15.45:5080>;tag=rj2nc38qem To: ;tag=j4X0e04aZjU2K Call-ID: joshempnx2kuh7smrt3x CSeq: 1974 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 P-Asserted-Identity: "30201009" jssip-0.2.1.js:684 JsSIP | DIALOG | New UAC dialog created: 1 jssip-0.2.1.js:2171 JsSIP | EVENT EMITTER | Emitting event: progress jssip-0.2.1.js:170 JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 Max-Forwards: 10 From: ;tag=rj2nc38qem To: < sip:30201009 at 116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: joshempnx2kuh7smrt3x CSeq: 1974 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Asserted-Identity: "30201009" < sip:30201009 at 116.214.15.45> ============================================================================= What am I doing wrong? Thanks, Paan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/daee9731/attachment-0001.html From paan.singh.coder at gmail.com Wed Jan 16 23:35:34 2013 From: paan.singh.coder at gmail.com (Paan Singh) Date: Thu, 17 Jan 2013 02:05:34 +0530 Subject: [Freeswitch-users] 488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH In-Reply-To: References: Message-ID: New SIP capture og with FreeSWITCH log syntax highlighting : http://pastebin.freeswitch.com/20457 On Thu, Jan 17, 2013 at 12:50 AM, Paan Singh wrote: > Hi, > > I'm using JsSIP from a webpage to make a SIP call to FS, using OverSIP as > a Websocket->SIP proxy. > > I'm able to Register successfully, but when I make a call from JsSIP UA to > FreeSWITCH, I get a 180 Ringing, but after that I get a 488 Not Acceptable > here.. Please help me figure out what I'm doing wrong here. > > > My inbound_codec_prefs is : > > > > Here is FreeSWITCH's log for the incoming call : > > ============================================= > port.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN > tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) > tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg 0x2913be0 > from (udp/116.214.15.45:5080) has 3113 bytes, veclen = 1 > tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x2913be0 (3113 > bytes) from udp/121.245.35.150:5080/sip next=(nil) > nta.c:2803 agent_recv_request() nta: received INVITE > sip:30201009 at 116.214.15.45 SIP/2.0 (CSeq 2141) > nta.c:3161 agent_aliases() nta: canonizing sip:30201009 at 116.214.15.45with contact > nta.c:3002 agent_recv_request() nta: INVITE (2141) going to a default leg > nta.c:1348 set_timeout() nta: timer set to 200 ms > nua_server.c:102 nua_stack_process_request() nua: > nua_stack_process_request: entering > nua_stack.c:899 nh_create() nua: nh_create: entering > nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > soa.c:282 soa_clone() soa_clone(static::0x26122e0, 0x2606e20, > 0x7f91840c4c50) called > soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) > called > nta.c:4313 nta_leg_tcreate() nta_leg_tcreate(0x7f91840356d0) > soa.c:1302 soa_init_offer_answer() > soa_init_offer_answer(static::0x7f91840b25e0) called > soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7f91840b25e0, > (nil), 0x285676c, 2333) called > nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f91840c4c50): adding > session usage > tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ > 121.245.35.150:5065 > tport.c:4026 tport_resolve() tport_resolve addrinfo = 121.245.35.150:5065 > tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not found > by name UDP/121.245.35.150:5065 > tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 564 bytes of 564 to udp/ > 121.245.35.150:5065 > tport.c:3472 tport_send_msg() tport_vsend returned 564 > nta.c:6678 incoming_reply() nta: sent 100 Trying for INVITE (2141) > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_invite 100 > Trying > nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call > state changed: init -> received, received offer > soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7f91840b25e0, > [0x7f91aa98a5e8], [0x7f91aa98a5e0], [(nil)]) called > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state 100 > Trying > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > 2013-01-17 00:25:27.733506 [NOTICE] switch_channel.c:968 New Channel > sofia/external/1000 at 116.214.15.45:5080[4aec7fd8-600e-11e2-8706-1365ba01ad52] > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_NEW > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:433 > (sofia/external/1000 at 116.214.15.45:5080) State NEW > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5576 Channel sofia/external/ > 1000 at 116.214.15.45:5080 entering state [received][100] > 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5587 Remote SDP: > v=0 > o=- 2957375511 2 IN IP4 127.0.0.1 > s=- > t=0 0 > a=group:BUNDLE audio video > m=audio 40111 RTP/SAVPF 103 104 0 8 106 105 13 126 > c=IN IP4 121.245.35.150 > a=rtcp:40111 IN IP4 121.245.35.150 > a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host > generation 0 > a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host > generation 0 > a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx > generation 0 > a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx > generation 0 > a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host > generation 0 > a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host > generation 0 > a=ice-ufrag:qiKF0ErbxaXldKI0 > a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw > a=ice-options:google-ice > a=mid:audio > a=rtcp-mux > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > a=ssrc:4278252985 cname:fMHr4uVHPcpHFfGu > a=ssrc:4278252985 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx > a=ssrc:4278252985 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx00 > m=video 40111 RTP/SAVPF 100 101 102 > c=IN IP4 121.245.35.150 > a=rtcp:40111 IN IP4 121.245.35.150 > a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host > generation 0 > a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host > generation 0 > a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx > generation 0 > a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx > generation 0 > a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host > generation 0 > a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host > generation 0 > a=ice-ufrag:qiKF0ErbxaXldKI0 > a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw > a=ice-options:google-ice > a=mid:video > a=rtcp-mux > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 > a=rtpmap:100 VP8/90000 > a=rtpmap:101 red/90000 > a=rtpmap:102 ulpfec/90000 > a=ssrc:169275022 cname:fMHr4uVHPcpHFfGu > a=ssrc:169275022 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx > a=ssrc:169275022 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx10 > > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 > AES_CM_128_HMAC_SHA1_80 inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2] > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:3155 Set Local Key [1 > AES_CM_128_HMAC_SHA1_80 inline:i8vqmBeU4+/sjJBndCbEQ4G87gmpYeVnqxh8Xuzu] > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:5277 No 2833 in SDP. > Disable 2833 dtmf and switch to INFO > 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:2994 > (sofia/external/1000 at 116.214.15:45:5080) Callstate Change DOWN -> HANGUP > 2013-01-17 00:25:27.753511 [NOTICE] sofia.c:5878 Hangup sofia/external/ > 1000 at 116.214.15.45:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION] > 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:3017 Send signal > sofia/external/1000 at 116.214.15.45:5080 [KILL] > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_HANGUP > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 > (sofia/external/1000 at 116.214.15.45:5080) State HANGUP > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:503 Channel sofia/external/ > 1000 at 116.214.15.45:5080 hanging up, cause: INCOMPATIBLE_DESTINATION > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:633 Responding to INVITE > with: 488 > nua.c:879 nua_respond() nua: nua_respond: entering > nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_respond > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:48 > sofia/external/1000 at 116.214.15.45:5080 Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 > (sofia/external/1000 at 116.214.15.45:5080) State HANGUP going to sleep > nua_stack.c:573 nua_stack_signal() nua(0x7f91840c4c50): recv signal > r_respond 488 Not Acceptable Here > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:446 > (sofia/external/1000 at 116.214.15.45:5080) State Change CS_HANGUP -> > CS_REPORTING > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) > called > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_REPORTING > nua_session.c:2316 nua_invite_server_respond() nua: > nua_invite_server_respond: entering > soa.c:1214 soa_clear_remote_sdp() > soa_clear_remote_sdp(static::0x7f91840b25e0) called > tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ > 121.245.35.150:5065 > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 > (sofia/external/1000 at 116.214.15.45:5080) State REPORTING > tport.c:4026 tport_resolve() tport_resolve addrinfo = 121.245.35.150:5065 > tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not found > by name UDP/121.245.35.150:5065 > tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 784 bytes of 784 to udp/ > 121.245.35.150:5065 > tport.c:3472 tport_send_msg() tport_vsend returned 784 > nta.c:6678 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE > (2141) > nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f91840c4c50): > removing session usage > nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call > state changed: received -> terminated > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state 488 > Not Acceptable Here > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_terminated > 488 Not Acceptable Here > soa.c:358 soa_destroy() soa_destroy(static::0x7f91840b25e0) called > nta.c:4366 nta_leg_destroy() nta_leg_destroy(0x7f91840356d0) > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering > nua_stack.c:569 nua_stack_signal() nua(0x7f91840c4c50): recv signal > r_destroy > nta.c:4366 nta_leg_destroy() nta_leg_destroy((nil)) > nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_destroy > 2013-01-17 00:25:27.753511 [DEBUG] mod_cdr_sqlite.c:102 Writing SQL to DB: > INSERT INTO cdr VALUES ("1000","1000","30201009","public","2013-01-17 > 00:25:27","","2013-01-17 > 00:25:27",0,0,"INCOMPATIBLE_DESTINATION","4aec7fd8-600e-11e2-8706-1365ba01ad52","","") > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:92 > sofia/external/1000 at 116.214.15.45:5080 Standard REPORTING, cause: > INCOMPATIBLE_DESTINATION > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 > (sofia/external/1000 at 116.214.15.45:5080) State REPORTING going to sleep > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/1000 at 116.214.15.45:5080) State Change CS_REPORTING -> > CS_DESTROY > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1499 Session 1794 > (sofia/external/1000 at 116.214.15.45:5080) Locked, Waiting on external > entities > 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1517 Session > 1794 (sofia/external/1000 at 116.214.15.45:5080) Ended > 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1521 Close > Channel sofia/external/1000 at 116.214.15.45:5080 [CS_DESTROY] > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:556 > (sofia/external/1000 at 116.214.15.45:5080) Callstate Change HANGUP -> DOWN > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:559 > (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_DESTROY > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 > (sofia/external/1000 at 116.214.15.45:5080) State DESTROY > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:396 sofia/external/ > 1000 at 116.214.15.45:5080 SOFIA DESTROY > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:99 > sofia/external/1000 at 116.214.15.45:5080 Standard DESTROY > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 > (sofia/external/1000 at 116.214.15.45:5080) State DESTROY going to sleep > nta.c:1294 agent_timer() nta: timer set next to 322 ms > tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN > tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) > tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg > 0x7f9198220070 from (udp/116.214.15.45:5080) has 339 bytes, veclen = 1 > tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x7f9198220070 > (339 bytes) from udp/121.245.35.150:5080/sip next=(nil) > nta.c:2803 agent_recv_request() nta: received ACK > sip:30201009 at 116.214.15.45 SIP/2.0 (CSeq 2141) > nta.c:2938 agent_recv_request() nta: ACK (2141) is going to INVITE (2141) > nta.c:1294 agent_timer() nta: timer set next to 4823 ms > nta.c:7010 _nta_incoming_timer() nta: timer I fired, terminate 488 response > nta.c:5718 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), > 0x7f91aa98ac20) > nta.c:7061 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, > 1/1 term, 1/1 free > nta.c:1279 agent_timer() nta: timer not set > ================================================================ > > Here is the Log from JsSIP in the browser that receives messages from > FreeSWITCH : > =========================================================== > JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 180 Ringing > Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 Record-Route: > Record-Route: > From: < > sip:1000 at 116.214.15.45:5080>;tag=rj2nc38qem To: < > sip:30201009 at 116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: > joshempnx2kuh7smrt3x CSeq: 1974 INVITE Contact: > User-Agent: > FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Accept: > application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, > UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, > replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 > P-Asserted-Identity: "30201009" > jssip-0.2.1.js:684 > JsSIP | DIALOG | New UAC dialog created: 1 jssip-0.2.1.js:2171 > JsSIP | EVENT EMITTER | Emitting event: progress jssip-0.2.1.js:170 > JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 488 Not > Acceptable Here Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 > Max-Forwards: 10 From: ;tag=rj2nc38qem To: < > sip:30201009 at 116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: > joshempnx2kuh7smrt3x CSeq: 1974 INVITE User-Agent: > FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Allow: INVITE, > ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces Allow-Events: talk, hold, > conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > Content-Length: 0 P-Asserted-Identity: "30201009" < > sip:30201009 at 116.214.15.45> > ============================================================================= > > > > > > > What am I doing wrong? Thanks, > Paan > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/6cad1d6f/attachment-0001.html From anthony.minessale at gmail.com Thu Jan 17 04:12:58 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Jan 2013 19:12:58 -0600 Subject: [Freeswitch-users] limit memory usage of freeswitch In-Reply-To: References: Message-ID: It does not eat memory.... if you attack it with sipp you may cause the internal pools to swell up but normal usage only consumes memory based on usage. On Wed, Jan 16, 2013 at 4:09 AM, jay binks wrote: > OpenVZ / Proxmox will allow you use set quota's and containerize your > freeswitch... > thats ONE way to do what your after. > > > On 16 January 2013 19:48, rentmycoder rentmycoder wrote: > >> Hi All, >> >> According to my experineces and some info found on FS lists it is the >> normal behaviour that freeswitch eats up all of the available memory... >> It's not a memory leak, but the standard behaviour of FS by design... >> Correct me if I'm wrong... >> >> Is there any way to limit FS memory usage??? >> >> I need another services to run on the FS box... >> >> Thanks, >> John >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/7d389dde/attachment.html From msc at freeswitch.org Thu Jan 17 04:17:31 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Jan 2013 17:17:31 -0800 Subject: [Freeswitch-users] Unloop extension In-Reply-To: <20130116100914.d57f06c7@mail.btcom.kz> References: <20130116100914.d57f06c7@mail.btcom.kz> Message-ID: No, it's available globally, whether used with $${} or ${} notation. You can also see it from fs_cli: freeswitch at default> global_getvar unroll_loops true More info about $${} vs. ${} is found here . -MC On Tue, Jan 15, 2013 at 8:09 PM, ????? ??????? wrote: > Ain't what was set with used in $${...} notation? > > ----- Original Message ----- > From: Michael Collins [mailto:msc at freeswitch.org] > To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > Sent: Wed, 16 Jan 2013 06:54:12 +0600 > Subject: Re: [Freeswitch-users] Unloop extension > > > > Yes, it's okay. The pre-process does a set and ${unroll_loops} gets set > as > > needed. Then this dp entry does the work: > > > > > > > > > > > > > > > > More info on sip_looped_call var: > > http://wiki.freeswitch.org/wiki/Variable_sip_looped_call > > > > -MC > > > > On Mon, Jan 14, 2013 at 9:33 PM, ????? ??????? > > wrote: > > > > > What unloop extension is used for? BTW: vars.xml pre-(process)-sets > > > unroll_loops, whereas DPs (public and default) are using channel > variable > > > syntax? Is this ok? > > > > > > Y. > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/c32ca31b/attachment.html From msc at freeswitch.org Thu Jan 17 04:18:56 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Jan 2013 17:18:56 -0800 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: Can you get a FS console log of this occurring? Also, what are the endpoints that are being connected? -MC On Wed, Jan 16, 2013 at 4:59 AM, Subhash wrote: > Hi all, > > I have a freeswitch configured with freetdm and wanpipe.My problem is with > dtmf. When calling from an external pstn number to freeswitch, dtmf seems > to be detected properly and enqueued it in the channel consistently. But > when call came to same channel again, dtmf which is stored in the > buffer for the previous call is used in the current call.So when i am done > with previous call i want to dequeue the dtmf in the channel .So the > problem seems to be something I've misconfigured with the Sangoma card. > Can anyone point me in the right direction? > > Thanks, > Subhash. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/5c4154c2/attachment.html From msc at freeswitch.org Thu Jan 17 04:27:58 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Jan 2013 17:27:58 -0800 Subject: [Freeswitch-users] conference-flags In-Reply-To: <07101B0D-F9A9-4845-A9B8-5F591A9F8949@kavun.ch> References: <68031E41-1E74-49C5-AC80-2ABFCBB1DB52@kavun.ch> <07101B0D-F9A9-4845-A9B8-5F591A9F8949@kavun.ch> Message-ID: I believe that these have to do with which video stream gets displayed. We have a user (sorry, can't recall name off top of my head) who does work with deaf/hearing-impaired and he needs to do video conferences where the translator does audio between the speaker and video to the deaf person. I think these affect that. I someone can shed some light on this that would be helpful. It's the third Monday of the week for me and I'm a little fried... Thanks, MC On Wed, Jan 16, 2013 at 7:41 AM, Emrah wrote: > Hi guys, > > If any of you know the answer to this I would be very grateful if you > could take a minute and post it on the Wiki or reply to my email. > > Thanks! > On Jan 11, 2013, at 2:28 PM, Emrah wrote: > > > Hi all, > > > > Where can I find some explanations on how to use conference-flags? > > > > How do you use video-bridge and video-floor and what do they do? > > > > Thanks for any pointers. > > > > E > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/310f239e/attachment-0001.html From support at ecn.net.au Thu Jan 17 05:58:31 2013 From: support at ecn.net.au (support at ecn.net.au) Date: Thu, 17 Jan 2013 12:58:31 +1000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting>, Message-ID: <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> Hi This sounds quite posible. I've tcpdumped sip headers on both the freeswitch and the backend PBX. The only significant difference I can see between our old SBC and FS is that the old SBC asserted a media attribute fmtp:18 annexb=no where as FS doesn't forward this. To test it out we have set dtmf_mode to none on both contexts (the context facing the telco and the context facing the pbx), and set (this may not be needed). then in the dialplan we have set prior to the bridge. -- When we sip trace however we are not appending the attribute to the SDP on the INVITE. Logs and sip headers below: EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060 export(sip_append_audio_sdp=a=fmtp:18 annexb=no) 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060 bridge(sofia/external/0737111111 at A.B.C.X:5060) 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1089 sofia/aapt/0731111111 at X.Y.Z.A:5060 EXPORTING[export_vars] [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to eve nt 2013-01-17 12:34:19.702102 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-01-17 12:34:19.702102 [NOTICE] switch_channel.c:968 New Channel sofia/external/0737111111 at A.B.C.X:5060 [a30c650a-87d3-4027-9010-213547d698aa] 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:4977 (sofia/external/0737111111 at A.B.C.X:5060) State Change CS_NEW -> CS_INIT 2013-01-17 12:34:19.702102 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/0737111111 at A.B.C.X:5060 [BREAK] 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0737111111 at A.B.C.X:5060) Running State Change CS_INIT 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:454 (sofia/external/0737111111 at A.B.C.X:5060) State INIT 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:86 sofia/external/0737111111 at A.B.C.X:5060 SOFIA INIT 2013-01-17 12:34:19.702102 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D s=FreeSWITCH c=IN IP4 A.B.C.D t=0 0 m=audio 24552 RTP/AVP 8 0 3 13 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv But then the actual Invite doesn't contain the attribute in the SDP send 1036 bytes to udp/[A.B.C.X]:5060 at 02:34:19.718385: ------------------------------------------------------------------------ INVITE sip:0737111111 at A.B.C.X:5060 SIP/2.0 Via: SIP/2.0/UDP A.B.C.D:5080;rport;branch=z9hG4bKpUvH6K1ppUSNp Max-Forwards: 69 From: "0731111111" ;tag=mH93ZHryv4H2g To: Call-ID: 3ca1df3f-daf1-1230-f389-002219a7c712 CSeq: 38858965 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.5.3+git~20121229T001759Z~e04eab7902 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 155 X-Nortel-Profile: DEFAULT X-FS-Support: update_display,send_info Remote-Party-ID: "0731111111" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D s=FreeSWITCH c=IN IP4 A.B.C.D t=0 0 m=audio 24552 RTP/AVP 8 0 3 13 a=ptime:20 ------------------------------------------------------------------------ Are we missing something do you think? Kind Regards, ________________________________ From: Steven Ayre [steveayre at gmail.com] Sent: Thursday, 17 January 2013 4:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SBC In-band DTMF However under freeswitch if we don't start_dtmf before the bridge the backend PBX boxes don't recognise the DTMF inband (even though the tones are audible ie you can hear them on a call recording on the PBX). Have we missed something here? We would have thought with inband DTMF on non compressed codec (no transcoding) that the tones would just work with the media stream? start_dtmf will detect inband DTMF, and then send out of band on the outgoing leg of the bridge. This is expected. You'll will need to call start_dtmf from dialplan, which can easily be done from dialplan by checking for some condition identifying calls requiring it. If they're authenticating to FS then it'd be trivial to set a variable in their user directory entry that you can test in the dialplan. Now this is just a guess... During the codec negotiation FS will tell your PBX that it supports telephone-event (RFC2833). It may be that because the PBX sees that in the SDP that it doesn't look for inband DTMF, while when it received the SDP direct from the customer it didn't contain telephone-event and so did check for inband DTMF. Indeed it'd probably be a good idea for them to do so, since if you receive the same DTMF digit both inband and through RFC2833 then you'd be duplicating digits. So my guess is that the PBX doesn't bother check for inband because FS tells it it's sending out-of-band, which'd differ from before. -Steve -Steve On 16 January 2013 04:02, support at ecn.net.au > wrote: > Hi All > > > > We're quite new to Freeswitch and are in the process of migrating from > OpenSer (as an SBC) to Freeswitch. > > > > Mostly all is working well, except an oddity on DTMF. > > > > Our scenario: > > > > Telco/SIP Provider A is passing us calls using DTMF inband. > > > > We have a freeswitch configured as a SBC using 2 sip profiles (telco and > internal) to topology hide and manage > > distribution of calls to the PBX servers located behind the SBC. > > > > The freeswitch will be handling up to a few hundred calls so we're trying to > keep it lightweight. > > > > Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling > customer needs. > > > > An example inbound call profile looks like this: > > > > > > > > > > > > data="sofia/external/123456 at INTERNAL.PBX.IP:5060"/> > > > > > > > > Initially when calling into the platform IVR type applications runinng on > our PBX boxes would not > > work (you could hear the DTMF but the platform did not recognise the tones). > > > > We have had to add the appliation start_dtmf in order for Freeswitch to pass > the DTMF to the Asterisk > > PBX behind the SBC. Interestingly on our OpenSer platform we just proxied > the media (rtpproxy) with > > inband DTMF from the Telco and our PBX boxes recognised the inband DTMF > tones on the PBX platforms and > > IVR type applications just worked. > > > > However under freeswitch if we don't start_dtmf before the bridge the > backend PBX boxes don't recognise > > the DTMF inband (even though the tones are audible ie you can hear them on a > call recording on the > > PBX). > > > > Have we missed something here? We would have thought with inband DTMF on > non compressed codec (no > > transcoding) that the tones would just work with the media stream? > > > > We have confirmed both legs are PCMA and when using start_dtmf the first > second of the call is clipped. > > > > > > > > Kind Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/ad5eb736/attachment.html From jaybinks at gmail.com Thu Jan 17 07:16:50 2013 From: jaybinks at gmail.com (jay binks) Date: Thu, 17 Jan 2013 14:16:50 +1000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> Message-ID: the problem with doing the sip traces how you have, is that you dont have the RTP Stream. id be using tcpdump on your boxes, so you can see does the RTP ( that appears to be cut off ) even get to your FS node. http://wiki.freeswitch.org/wiki/Packet_Capture do something like tcpdump -nq -s 0 -i eth0 -w /tmp/dump.pcap On 17 January 2013 12:58, support at ecn.net.au wrote: > Hi > > This sounds quite posible. > > I've tcpdumped sip headers on both the freeswitch and the backend PBX. > > The only significant difference I can see between our old SBC and FS is > that the old SBC > asserted a media attribute fmtp:18 annexb=no where as FS doesn't forward > this. > > To test it out we have set dtmf_mode to none on both contexts (the context > facing the telco > and the context facing the pbx), and set > > > > (this may not be needed). > > then in the dialplan we have > > > > set prior to the bridge. > > -- > > When we sip trace however we are not appending the attribute to the SDP on > the INVITE. > > Logs and sip headers below: > > > EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060export(sip_append_audio_sdp=a=fmtp:18 annexb=no) > 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1135 EXPORT > (export_vars) [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] > EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060 bridge( > sofia/external/0737111111 at A.B.C.X:5060) > 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1089 > sofia/aapt/0731111111 at X.Y.Z.A:5060 EXPORTING[export_vars] > [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to eve > nt > 2013-01-17 12:34:19.702102 [DEBUG] switch_ivr_originate.c:2022 Parsing > global variables > 2013-01-17 12:34:19.702102 [NOTICE] switch_channel.c:968 New Channel > sofia/external/0737111111 at A.B.C.X:5060[a30c650a-87d3-4027-9010-213547d698aa] > 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:4977 ( > sofia/external/0737111111 at A.B.C.X:5060) State Change CS_NEW -> CS_INIT > 2013-01-17 12:34:19.702102 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/0737111111 at A.B.C.X:5060 [BREAK] > 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:415 ( > sofia/external/0737111111 at A.B.C.X:5060) Running State Change CS_INIT > 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:454 ( > sofia/external/0737111111 at A.B.C.X:5060) State INIT > 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:86 > sofia/external/0737111111 at A.B.C.X:5060 SOFIA INIT > 2013-01-17 12:34:19.702102 [DEBUG] sofia_glue.c:2647 Local SDP: > v=0 > o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D > s=FreeSWITCH > c=IN IP4 A.B.C.D > t=0 0 > m=audio 24552 RTP/AVP 8 0 3 13 > a=fmtp:18 annexb=no > a=ptime:20 > a=sendrecv > > > But then the actual Invite doesn't contain the attribute in the SDP > > send 1036 bytes to udp/[A.B.C.X]:5060 at 02:34:19.718385: > ------------------------------------------------------------------------ > INVITE sip:0737111111 at A.B.C.X:5060 SIP/2.0 > Via: SIP/2.0/UDP A.B.C.D:5080;rport;branch=z9hG4bKpUvH6K1ppUSNp > Max-Forwards: 69 > From: "0731111111" ;tag=mH93ZHryv4H2g > To: > Call-ID: 3ca1df3f-daf1-1230-f389-002219a7c712 > CSeq: 38858965 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.2.5.3 > +git~20121229T001759Z~e04eab7902 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 155 > X-Nortel-Profile: DEFAULT > X-FS-Support: update_display,send_info > Remote-Party-ID: "0731111111" >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D > s=FreeSWITCH > c=IN IP4 A.B.C.D > t=0 0 > m=audio 24552 RTP/AVP 8 0 3 13 > a=ptime:20 > ------------------------------------------------------------------------ > > > > Are we missing something do you think? > > > > Kind Regards, > > ------------------------------ > *From:* Steven Ayre [steveayre at gmail.com] > *Sent:* Thursday, 17 January 2013 4:14 AM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SBC In-band DTMF > > However under freeswitch if we don't start_dtmf before the bridge the >> backend PBX boxes don't recognise >> the DTMF inband (even though the tones are audible ie you can hear them >> on a call recording on the >> PBX). >> >> Have we missed something here? We would have thought with inband DTMF on >> non compressed codec (no >> transcoding) that the tones would just work with the media stream? > > > start_dtmf will detect inband DTMF, and then send out of band on the > outgoing leg of the bridge. This is expected. You'll will need to call > start_dtmf from dialplan, which can easily be done from dialplan by > checking for some condition identifying calls requiring it. If they're > authenticating to FS then it'd be trivial to set a variable in their user > directory entry that you can test in the dialplan. > > Now this is just a guess... During the codec negotiation FS will tell > your PBX that it supports telephone-event (RFC2833). It may be that because > the PBX sees that in the SDP that it doesn't look for inband DTMF, while > when it received the SDP direct from the customer it didn't contain > telephone-event and so did check for inband DTMF. Indeed it'd probably be a > good idea for them to do so, since if you receive the same DTMF digit both > inband and through RFC2833 then you'd be duplicating digits. So my guess > is that the PBX doesn't bother check for inband because FS tells it it's > sending out-of-band, which'd differ from before. > > -Steve > > > > -Steve > > > > > On 16 January 2013 04:02, support at ecn.net.au wrote: > > Hi All > > > > > > > > We're quite new to Freeswitch and are in the process of migrating from > > OpenSer (as an SBC) to Freeswitch. > > > > > > > > Mostly all is working well, except an oddity on DTMF. > > > > > > > > Our scenario: > > > > > > > > Telco/SIP Provider A is passing us calls using DTMF inband. > > > > > > > > We have a freeswitch configured as a SBC using 2 sip profiles (telco and > > internal) to topology hide and manage > > > > distribution of calls to the PBX servers located behind the SBC. > > > > > > > > The freeswitch will be handling up to a few hundred calls so we're > trying to > > keep it lightweight. > > > > > > > > Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling > > customer needs. > > > > > > > > An example inbound call profile looks like this: > > > > > > > > > > > > > > > > > > > > > > > > > data="sofia/external/123456 at INTERNAL.PBX.IP:5060"/> > > > > > > > > > > > > > > > > Initially when calling into the platform IVR type applications runinng on > > our PBX boxes would not > > > > work (you could hear the DTMF but the platform did not recognise the > tones). > > > > > > > > We have had to add the appliation start_dtmf in order for Freeswitch to > pass > > the DTMF to the Asterisk > > > > PBX behind the SBC. Interestingly on our OpenSer platform we just > proxied > > the media (rtpproxy) with > > > > inband DTMF from the Telco and our PBX boxes recognised the inband DTMF > > tones on the PBX platforms and > > > > IVR type applications just worked. > > > > > > > > However under freeswitch if we don't start_dtmf before the bridge the > > backend PBX boxes don't recognise > > > > the DTMF inband (even though the tones are audible ie you can hear them > on a > > call recording on the > > > > PBX). > > > > > > > > Have we missed something here? We would have thought with inband DTMF on > > non compressed codec (no > > > > transcoding) that the tones would just work with the media stream? > > > > > > > > We have confirmed both legs are PCMA and when using start_dtmf the first > > second of the call is clipped. > > > > > > > > > > > > > > > > Kind Regards, > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/ffe427dd/attachment-0001.html From support at ecn.net.au Thu Jan 17 07:30:01 2013 From: support at ecn.net.au (support at ecn.net.au) Date: Thu, 17 Jan 2013 14:30:01 +1000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting>, Message-ID: <8A80C99638A5E24E83696EAE33A9B6B968B5302806@twbexmbx001.ecn.hosting> We've tcpdumped the data and exported this from wireshark. The dialplan for this b-leg of the bridged call is: > 10.0.0.1 was the Freeswitch, 10.0.0.2 is the backend PBX. tcpdump SIP capture on the backend PBX for the invite from the Freeswitch SBC. (Note the SDP does not contain the media attribute for fmtp). INVITE sip:0737370000 at 10.0.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1:5080;rport;branch=z9hG4bKZc0gp7rpvB2Xe Max-Forwards: 69 From: "0731050000" ;tag=QcNe52a9KZmtK To: Call-ID: a4944ebb-dafe-1230-f389-002219a7c712 CSeq: 38861844 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.5.3+git~20121229T001759Z~e04eab7902 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 155 X-Nortel-Profile: DEFAULT X-FS-Support: update_display,send_info Remote-Party-ID: "0731050000" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1358367831 1358367832 IN IP4 10.0.0.1 s=FreeSWITCH c=IN IP4 10.0.0.1 t=0 0 m=audio 27986 RTP/AVP 8 0 3 13 a=ptime:20 Kind Regards, ________________________________ From: jay binks [jaybinks at gmail.com] Sent: Thursday, 17 January 2013 2:16 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SBC In-band DTMF the problem with doing the sip traces how you have, is that you dont have the RTP Stream. id be using tcpdump on your boxes, so you can see does the RTP ( that appears to be cut off ) even get to your FS node. http://wiki.freeswitch.org/wiki/Packet_Capture do something like tcpdump -nq -s 0 -i eth0 -w /tmp/dump.pcap On 17 January 2013 12:58, support at ecn.net.au > wrote: Hi This sounds quite posible. I've tcpdumped sip headers on both the freeswitch and the backend PBX. The only significant difference I can see between our old SBC and FS is that the old SBC asserted a media attribute fmtp:18 annexb=no where as FS doesn't forward this. To test it out we have set dtmf_mode to none on both contexts (the context facing the telco and the context facing the pbx), and set (this may not be needed). then in the dialplan we have set prior to the bridge. -- When we sip trace however we are not appending the attribute to the SDP on the INVITE. Logs and sip headers below: EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060 export(sip_append_audio_sdp=a=fmtp:18 annexb=no) 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060 bridge(sofia/external/0737111111 at A.B.C.X:5060) 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1089 sofia/aapt/0731111111 at X.Y.Z.A:5060 EXPORTING[export_vars] [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to eve nt 2013-01-17 12:34:19.702102 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-01-17 12:34:19.702102 [NOTICE] switch_channel.c:968 New Channel sofia/external/0737111111 at A.B.C.X:5060 [a30c650a-87d3-4027-9010-213547d698aa] 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:4977 (sofia/external/0737111111 at A.B.C.X:5060) State Change CS_NEW -> CS_INIT 2013-01-17 12:34:19.702102 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/0737111111 at A.B.C.X:5060 [BREAK] 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0737111111 at A.B.C.X:5060) Running State Change CS_INIT 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:454 (sofia/external/0737111111 at A.B.C.X:5060) State INIT 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:86 sofia/external/0737111111 at A.B.C.X:5060 SOFIA INIT 2013-01-17 12:34:19.702102 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D s=FreeSWITCH c=IN IP4 A.B.C.D t=0 0 m=audio 24552 RTP/AVP 8 0 3 13 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv But then the actual Invite doesn't contain the attribute in the SDP send 1036 bytes to udp/[A.B.C.X]:5060 at 02:34:19.718385: ------------------------------------------------------------------------ INVITE sip:0737111111 at A.B.C.X:5060 SIP/2.0 Via: SIP/2.0/UDP A.B.C.D:5080;rport;branch=z9hG4bKpUvH6K1ppUSNp Max-Forwards: 69 From: "0731111111" ;tag=mH93ZHryv4H2g To: Call-ID: 3ca1df3f-daf1-1230-f389-002219a7c712 CSeq: 38858965 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.5.3+git~20121229T001759Z~e04eab7902 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 155 X-Nortel-Profile: DEFAULT X-FS-Support: update_display,send_info Remote-Party-ID: "0731111111" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D s=FreeSWITCH c=IN IP4 A.B.C.D t=0 0 m=audio 24552 RTP/AVP 8 0 3 13 a=ptime:20 ------------------------------------------------------------------------ Are we missing something do you think? Kind Regards, ________________________________ From: Steven Ayre [steveayre at gmail.com] Sent: Thursday, 17 January 2013 4:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SBC In-band DTMF However under freeswitch if we don't start_dtmf before the bridge the backend PBX boxes don't recognise the DTMF inband (even though the tones are audible ie you can hear them on a call recording on the PBX). Have we missed something here? We would have thought with inband DTMF on non compressed codec (no transcoding) that the tones would just work with the media stream? start_dtmf will detect inband DTMF, and then send out of band on the outgoing leg of the bridge. This is expected. You'll will need to call start_dtmf from dialplan, which can easily be done from dialplan by checking for some condition identifying calls requiring it. If they're authenticating to FS then it'd be trivial to set a variable in their user directory entry that you can test in the dialplan. Now this is just a guess... During the codec negotiation FS will tell your PBX that it supports telephone-event (RFC2833). It may be that because the PBX sees that in the SDP that it doesn't look for inband DTMF, while when it received the SDP direct from the customer it didn't contain telephone-event and so did check for inband DTMF. Indeed it'd probably be a good idea for them to do so, since if you receive the same DTMF digit both inband and through RFC2833 then you'd be duplicating digits. So my guess is that the PBX doesn't bother check for inband because FS tells it it's sending out-of-band, which'd differ from before. -Steve -Steve On 16 January 2013 04:02, support at ecn.net.au > wrote: > Hi All > > > > We're quite new to Freeswitch and are in the process of migrating from > OpenSer (as an SBC) to Freeswitch. > > > > Mostly all is working well, except an oddity on DTMF. > > > > Our scenario: > > > > Telco/SIP Provider A is passing us calls using DTMF inband. > > > > We have a freeswitch configured as a SBC using 2 sip profiles (telco and > internal) to topology hide and manage > > distribution of calls to the PBX servers located behind the SBC. > > > > The freeswitch will be handling up to a few hundred calls so we're trying to > keep it lightweight. > > > > Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling > customer needs. > > > > An example inbound call profile looks like this: > > > > > > > > > > > > data="sofia/external/123456 at INTERNAL.PBX.IP:5060"/> > > > > > > > > Initially when calling into the platform IVR type applications runinng on > our PBX boxes would not > > work (you could hear the DTMF but the platform did not recognise the tones). > > > > We have had to add the appliation start_dtmf in order for Freeswitch to pass > the DTMF to the Asterisk > > PBX behind the SBC. Interestingly on our OpenSer platform we just proxied > the media (rtpproxy) with > > inband DTMF from the Telco and our PBX boxes recognised the inband DTMF > tones on the PBX platforms and > > IVR type applications just worked. > > > > However under freeswitch if we don't start_dtmf before the bridge the > backend PBX boxes don't recognise > > the DTMF inband (even though the tones are audible ie you can hear them on a > call recording on the > > PBX). > > > > Have we missed something here? We would have thought with inband DTMF on > non compressed codec (no > > transcoding) that the tones would just work with the media stream? > > > > We have confirmed both legs are PCMA and when using start_dtmf the first > second of the call is clipped. > > > > > > > > Kind Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/eb40f434/attachment-0001.html From valernur at yahoo.com Thu Jan 17 09:29:59 2013 From: valernur at yahoo.com (Valer Nur) Date: Wed, 16 Jan 2013 22:29:59 -0800 (PST) Subject: [Freeswitch-users] Mod_Conference Improvement (Denoise) In-Reply-To: References: <5274F7F1-8BC9-4721-B98F-A163495CD730@jerris.com> Message-ID: <1358404199.68203.YahooMailNeo@web162901.mail.bf1.yahoo.com> I think the best way to solve this is by doing AGC first. This should make sure audio level is similar for all participants. ________________________________ From: Anthony Minessale To: FreeSWITCH Users Help Sent: Wednesday, January 16, 2013 6:23 PM Subject: Re: [Freeswitch-users] Mod_Conference Improvement (Denoise) There is also the preprocess app which uses a media bug in the core for a variety of things but is incomplete it's using a bit of code from speex so we can?certainly improve that as well. ? On Tue, Jan 15, 2013 at 8:58 PM, Michael Jerris wrote: If you wanted to implement something like this you could do so as a media bug, it does not need to touch mod_conference in any way. ?Take a look at mod_ladspa for an example of a similar module that modifies audio using a media bug. ?Give it a try and see if you can get something working. > > >Mike > > >On Jan 15, 2013, at 8:03 PM, Michael Collins wrote: > >While I don't have a problem with the concept of noise filtering I have to point out that each member of the conference can have his or her own noise level setting. From fs_cli: >> >>conference energy [] >> >>You only need to apply it to the person who is in the noisy location. In fact, in the example configuration the user can dial 9 to increase the energy threshold (for when he's in a noisy environment) or press 7 to decrease the threshold. Pressing 8 will reset the energy to normal, which I believe is 200. >> >>-MC >> >> >>On Tue, Jan 15, 2013 at 4:24 PM, Usama Zaidi wrote: >> >>Hi, >>> >>>I talked about this a while back on the channel, the way how energy-level is implemented in mod_conference isn't the right way to handle noise in my opinion because the participants that get bridged would have a different noise floor depending on where they are located, so if I set the value of energy-level too high to accommodate one participant, any other participant in a quite sitting would never get bridged, I proposed we should tackle this the way Asterisk does using a denoise dialplan function (https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE), we can use libav for noise filtering on the muxed audio. I'm willing to contribute if one of the devs would help me out regarding implementation. Any comments? >>> >>>-Regards >>> >>> >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/0a791d57/attachment.html From william.king at quentustech.com Thu Jan 17 09:37:19 2013 From: william.king at quentustech.com (William King) Date: Wed, 16 Jan 2013 22:37:19 -0800 Subject: [Freeswitch-users] Mod_Conference Improvement (Denoise) In-Reply-To: <1358404199.68203.YahooMailNeo@web162901.mail.bf1.yahoo.com> References: <5274F7F1-8BC9-4721-B98F-A163495CD730@jerris.com> <1358404199.68203.YahooMailNeo@web162901.mail.bf1.yahoo.com> Message-ID: <50F79C1F.8040804@quentustech.com> That feature already exists, but appears to still be undocumented. Added just over a year ago. http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_23 William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 01/16/2013 10:29 PM, Valer Nur wrote: > I think the best way to solve this is by doing AGC first. This should > make sure audio level is similar for all participants. > > > ------------------------------------------------------------------------ > *From:* Anthony Minessale > *To:* FreeSWITCH Users Help > *Sent:* Wednesday, January 16, 2013 6:23 PM > *Subject:* Re: [Freeswitch-users] Mod_Conference Improvement (Denoise) > > There is also the preprocess app which uses a media bug in the core for > a variety of things but is incomplete it's using a bit of code from > speex so we can certainly improve that as well. > > > > > On Tue, Jan 15, 2013 at 8:58 PM, Michael Jerris > wrote: > > If you wanted to implement something like this you could do so as a > media bug, it does not need to touch mod_conference in any way. > Take a look at mod_ladspa for an example of a similar module that > modifies audio using a media bug. Give it a try and see if you can > get something working. > > Mike > > On Jan 15, 2013, at 8:03 PM, Michael Collins > wrote: > >> While I don't have a problem with the concept of noise filtering I >> have to point out that each member of the conference can have his >> or her own noise level setting. From fs_cli: >> >> conference energy >> [] >> >> You only need to apply it to the person who is in the noisy >> location. In fact, in the example configuration the user can dial >> 9 to increase the energy threshold (for when he's in a noisy >> environment) or press 7 to decrease the threshold. Pressing 8 will >> reset the energy to normal, which I believe is 200. >> >> -MC >> >> On Tue, Jan 15, 2013 at 4:24 PM, Usama Zaidi > > wrote: >> >> Hi, >> >> I talked about this a while back on the channel, the way how >> energy-level is implemented in mod_conference isn't the right >> way to handle noise in my opinion because the participants >> that get bridged would have a different noise floor depending >> on where they are located, so if I set the value of >> energy-level too high to accommodate one participant, any >> other participant in a quite sitting would never get bridged, >> I proposed we should tackle this the way Asterisk does using a >> denoise dialplan function >> (https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE), >> we can use libav for noise filtering on the muxed audio. I'm >> willing to contribute if one of the devs would help me out >> regarding implementation. Any comments? >> >> -Regards >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From royj at yandex.ru Thu Jan 17 10:24:25 2013 From: royj at yandex.ru (royj) Date: Thu, 17 Jan 2013 11:24:25 +0400 Subject: [Freeswitch-users] 488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH In-Reply-To: References: Message-ID: <20130117112425.5ba534597f72c69dc122e214@yandex.ru> http://jira.freeswitch.org/browse/FS-4606 On Thu, 17 Jan 2013 02:05:34 +0530 Paan Singh wrote: > New SIP capture og with FreeSWITCH log syntax highlighting : > http://pastebin.freeswitch.com/20457 > > > On Thu, Jan 17, 2013 at 12:50 AM, Paan Singh wrote: > > > Hi, > > > > I'm using JsSIP from a webpage to make a SIP call to FS, using OverSIP as > > a Websocket->SIP proxy. > > > > I'm able to Register successfully, but when I make a call from JsSIP UA to > > FreeSWITCH, I get a 180 Ringing, but after that I get a 488 Not Acceptable > > here.. Please help me figure out what I'm doing wrong here. > > > > > > My inbound_codec_prefs is : > > > > > > > > Here is FreeSWITCH's log for the incoming call : > > > > ============================================= > > port.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN > > tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) > > tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg 0x2913be0 > > from (udp/116.214.15.45:5080) has 3113 bytes, veclen = 1 > > tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x2913be0 (3113 > > bytes) from udp/121.245.35.150:5080/sip next=(nil) > > nta.c:2803 agent_recv_request() nta: received INVITE > > sip:30201009 at 116.214.15.45 SIP/2.0 (CSeq 2141) > > nta.c:3161 agent_aliases() nta: canonizing sip:30201009 at 116.214.15.45with contact > > nta.c:3002 agent_recv_request() nta: INVITE (2141) going to a default leg > > nta.c:1348 set_timeout() nta: timer set to 200 ms > > nua_server.c:102 nua_stack_process_request() nua: > > nua_stack_process_request: entering > > nua_stack.c:899 nh_create() nua: nh_create: entering > > nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering > > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > > soa.c:282 soa_clone() soa_clone(static::0x26122e0, 0x2606e20, > > 0x7f91840c4c50) called > > soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) > > called > > nta.c:4313 nta_leg_tcreate() nta_leg_tcreate(0x7f91840356d0) > > soa.c:1302 soa_init_offer_answer() > > soa_init_offer_answer(static::0x7f91840b25e0) called > > soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7f91840b25e0, > > (nil), 0x285676c, 2333) called > > nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f91840c4c50): adding > > session usage > > tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ > > 121.245.35.150:5065 > > tport.c:4026 tport_resolve() tport_resolve addrinfo = 121.245.35.150:5065 > > tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not found > > by name UDP/121.245.35.150:5065 > > tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 564 bytes of 564 to udp/ > > 121.245.35.150:5065 > > tport.c:3472 tport_send_msg() tport_vsend returned 564 > > nta.c:6678 incoming_reply() nta: sent 100 Trying for INVITE (2141) > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_invite 100 > > Trying > > nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call > > state changed: init -> received, received offer > > soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7f91840b25e0, > > [0x7f91aa98a5e8], [0x7f91aa98a5e0], [(nil)]) called > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state 100 > > Trying > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > entering > > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > > 2013-01-17 00:25:27.733506 [NOTICE] switch_channel.c:968 New Channel > > sofia/external/1000 at 116.214.15.45:5080[4aec7fd8-600e-11e2-8706-1365ba01ad52] > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send signal > > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > entering > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send signal > > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:415 > > (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_NEW > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:433 > > (sofia/external/1000 at 116.214.15.45:5080) State NEW > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5576 Channel sofia/external/ > > 1000 at 116.214.15.45:5080 entering state [received][100] > > 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5587 Remote SDP: > > v=0 > > o=- 2957375511 2 IN IP4 127.0.0.1 > > s=- > > t=0 0 > > a=group:BUNDLE audio video > > m=audio 40111 RTP/SAVPF 103 104 0 8 106 105 13 126 > > c=IN IP4 121.245.35.150 > > a=rtcp:40111 IN IP4 121.245.35.150 > > a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host > > generation 0 > > a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host > > generation 0 > > a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx > > generation 0 > > a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx > > generation 0 > > a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host > > generation 0 > > a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host > > generation 0 > > a=ice-ufrag:qiKF0ErbxaXldKI0 > > a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw > > a=ice-options:google-ice > > a=mid:audio > > a=rtcp-mux > > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > > inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 > > a=rtpmap:103 ISAC/16000 > > a=rtpmap:104 ISAC/32000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:106 CN/32000 > > a=rtpmap:105 CN/16000 > > a=rtpmap:13 CN/8000 > > a=rtpmap:126 telephone-event/8000 > > a=ssrc:4278252985 cname:fMHr4uVHPcpHFfGu > > a=ssrc:4278252985 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx > > a=ssrc:4278252985 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx00 > > m=video 40111 RTP/SAVPF 100 101 102 > > c=IN IP4 121.245.35.150 > > a=rtcp:40111 IN IP4 121.245.35.150 > > a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host > > generation 0 > > a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host > > generation 0 > > a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx > > generation 0 > > a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx > > generation 0 > > a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host > > generation 0 > > a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host > > generation 0 > > a=ice-ufrag:qiKF0ErbxaXldKI0 > > a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw > > a=ice-options:google-ice > > a=mid:video > > a=rtcp-mux > > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > > inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 > > a=rtpmap:100 VP8/90000 > > a=rtpmap:101 red/90000 > > a=rtpmap:102 ulpfec/90000 > > a=ssrc:169275022 cname:fMHr4uVHPcpHFfGu > > a=ssrc:169275022 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx > > a=ssrc:169275022 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx10 > > > > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 > > AES_CM_128_HMAC_SHA1_80 inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2] > > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:3155 Set Local Key [1 > > AES_CM_128_HMAC_SHA1_80 inline:i8vqmBeU4+/sjJBndCbEQ4G87gmpYeVnqxh8Xuzu] > > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:5277 No 2833 in SDP. > > Disable 2833 dtmf and switch to INFO > > 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:2994 > > (sofia/external/1000 at 116.214.15:45:5080) Callstate Change DOWN -> HANGUP > > 2013-01-17 00:25:27.753511 [NOTICE] sofia.c:5878 Hangup sofia/external/ > > 1000 at 116.214.15.45:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION] > > 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:3017 Send signal > > sofia/external/1000 at 116.214.15.45:5080 [KILL] > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal > > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 > > (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_HANGUP > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 > > (sofia/external/1000 at 116.214.15.45:5080) State HANGUP > > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:503 Channel sofia/external/ > > 1000 at 116.214.15.45:5080 hanging up, cause: INCOMPATIBLE_DESTINATION > > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:633 Responding to INVITE > > with: 488 > > nua.c:879 nua_respond() nua: nua_respond: entering > > nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_respond > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:48 > > sofia/external/1000 at 116.214.15.45:5080 Standard HANGUP, cause: > > INCOMPATIBLE_DESTINATION > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 > > (sofia/external/1000 at 116.214.15.45:5080) State HANGUP going to sleep > > nua_stack.c:573 nua_stack_signal() nua(0x7f91840c4c50): recv signal > > r_respond 488 Not Acceptable Here > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:446 > > (sofia/external/1000 at 116.214.15.45:5080) State Change CS_HANGUP -> > > CS_REPORTING > > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal > > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > > soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) > > called > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 > > (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_REPORTING > > nua_session.c:2316 nua_invite_server_respond() nua: > > nua_invite_server_respond: entering > > soa.c:1214 soa_clear_remote_sdp() > > soa_clear_remote_sdp(static::0x7f91840b25e0) called > > tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ > > 121.245.35.150:5065 > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 > > (sofia/external/1000 at 116.214.15.45:5080) State REPORTING > > tport.c:4026 tport_resolve() tport_resolve addrinfo = 121.245.35.150:5065 > > tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not found > > by name UDP/121.245.35.150:5065 > > tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 784 bytes of 784 to udp/ > > 121.245.35.150:5065 > > tport.c:3472 tport_send_msg() tport_vsend returned 784 > > nta.c:6678 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE > > (2141) > > nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f91840c4c50): > > removing session usage > > nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call > > state changed: received -> terminated > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state 488 > > Not Acceptable Here > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_terminated > > 488 Not Acceptable Here > > soa.c:358 soa_destroy() soa_destroy(static::0x7f91840b25e0) called > > nta.c:4366 nta_leg_destroy() nta_leg_destroy(0x7f91840356d0) > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > entering > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > entering > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering > > nua_stack.c:569 nua_stack_signal() nua(0x7f91840c4c50): recv signal > > r_destroy > > nta.c:4366 nta_leg_destroy() nta_leg_destroy((nil)) > > nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_destroy > > 2013-01-17 00:25:27.753511 [DEBUG] mod_cdr_sqlite.c:102 Writing SQL to DB: > > INSERT INTO cdr VALUES ("1000","1000","30201009","public","2013-01-17 > > 00:25:27","","2013-01-17 > > 00:25:27",0,0,"INCOMPATIBLE_DESTINATION","4aec7fd8-600e-11e2-8706-1365ba01ad52","","") > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:92 > > sofia/external/1000 at 116.214.15.45:5080 Standard REPORTING, cause: > > INCOMPATIBLE_DESTINATION > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 > > (sofia/external/1000 at 116.214.15.45:5080) State REPORTING going to sleep > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:440 > > (sofia/external/1000 at 116.214.15.45:5080) State Change CS_REPORTING -> > > CS_DESTROY > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal > > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1499 Session 1794 > > (sofia/external/1000 at 116.214.15.45:5080) Locked, Waiting on external > > entities > > 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1517 Session > > 1794 (sofia/external/1000 at 116.214.15.45:5080) Ended > > 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1521 Close > > Channel sofia/external/1000 at 116.214.15.45:5080 [CS_DESTROY] > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:556 > > (sofia/external/1000 at 116.214.15.45:5080) Callstate Change HANGUP -> DOWN > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:559 > > (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_DESTROY > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 > > (sofia/external/1000 at 116.214.15.45:5080) State DESTROY > > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:396 sofia/external/ > > 1000 at 116.214.15.45:5080 SOFIA DESTROY > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:99 > > sofia/external/1000 at 116.214.15.45:5080 Standard DESTROY > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 > > (sofia/external/1000 at 116.214.15.45:5080) State DESTROY going to sleep > > nta.c:1294 agent_timer() nta: timer set next to 322 ms > > tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN > > tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) > > tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg > > 0x7f9198220070 from (udp/116.214.15.45:5080) has 339 bytes, veclen = 1 > > tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x7f9198220070 > > (339 bytes) from udp/121.245.35.150:5080/sip next=(nil) > > nta.c:2803 agent_recv_request() nta: received ACK > > sip:30201009 at 116.214.15.45 SIP/2.0 (CSeq 2141) > > nta.c:2938 agent_recv_request() nta: ACK (2141) is going to INVITE (2141) > > nta.c:1294 agent_timer() nta: timer set next to 4823 ms > > nta.c:7010 _nta_incoming_timer() nta: timer I fired, terminate 488 response > > nta.c:5718 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), > > 0x7f91aa98ac20) > > nta.c:7061 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, > > 1/1 term, 1/1 free > > nta.c:1279 agent_timer() nta: timer not set > > ================================================================ > > > > Here is the Log from JsSIP in the browser that receives messages from > > FreeSWITCH : > > =========================================================== > > JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 180 Ringing > > Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 Record-Route: > > Record-Route: > > From: < > > sip:1000 at 116.214.15.45:5080>;tag=rj2nc38qem To: < > > sip:30201009 at 116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: > > joshempnx2kuh7smrt3x CSeq: 1974 INVITE Contact: > > User-Agent: > > FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Accept: > > application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, > > UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, > > replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 > > P-Asserted-Identity: "30201009" > > jssip-0.2.1.js:684 > > JsSIP | DIALOG | New UAC dialog created: 1 jssip-0.2.1.js:2171 > > JsSIP | EVENT EMITTER | Emitting event: progress jssip-0.2.1.js:170 > > JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 488 Not > > Acceptable Here Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 > > Max-Forwards: 10 From: ;tag=rj2nc38qem To: < > > sip:30201009 at 116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: > > joshempnx2kuh7smrt3x CSeq: 1974 INVITE User-Agent: > > FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Allow: INVITE, > > ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY > > Supported: timer, precondition, path, replaces Allow-Events: talk, hold, > > conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > > Content-Length: 0 P-Asserted-Identity: "30201009" < > > sip:30201009 at 116.214.15.45> > > ============================================================================= > > > > > > > > > > > > > > What am I doing wrong? Thanks, > > Paan > > > > From mario_fs at mgtech.com Thu Jan 17 01:57:18 2013 From: mario_fs at mgtech.com (Mario G) Date: Wed, 16 Jan 2013 14:57:18 -0800 Subject: [Freeswitch-users] Freeswitch TLS and Yealink t26p In-Reply-To: <6C4DC695-57FA-4CA8-9332-CC19B974B85A@kavun.ch> References: <1356082510.23186.154.camel@marces.madrid.commsmundi.com> <1356090896.23186.170.camel@marces.madrid.commsmundi.com> <50E482F4.8060701@quentustech.com> <1358338707.5310.32.camel@vmmarces.vm.marces.com> <6C4DC695-57FA-4CA8-9332-CC19B974B85A@kavun.ch> Message-ID: <8DEB93D1-CD4C-4A26-9794-93F52CBA13D6@mgtech.com> I actually got back responses to wishlist and questions in 1-3 days at http://forum.yealink.com/forum/ Mario G On Jan 16, 2013, at 7:40 AM, Emrah wrote: > Sorry to hijack the thread, but where do you report bugs / feature requests? > I haven't been able to reach any knowledgeable person via their support email. > > Cheers and thank you > On Jan 16, 2013, at 7:18 AM, Antonio Silva wrote: > >> Hi, >> >> Nice to now. >> I didn't have this problem, i could upload a custom certificate and with the option "Only Accept Trusted Certificates" to enabled. >> >> I had another issue with ldap navigation, i report to them, they were quite fast to reply, and there is a new firmware 6.70.0.120 that solves the problem. >> >> Best regards, >> >> Ant?nio Silva >> >> >> On Wed, 2013-01-02 at 10:56 -0800, William King wrote: >>> Another thing to be aware of is that there is an outstanding bug where >>> Yealinks are not able to load certain custom CA's because the time on >>> the phone is not being synced to NTP before the CA is validated and loaded. >>> >>> Once you changed the transport have you run into any new issues? >>> >>> William King >>> Senior Engineer >>> Quentus Technologies, INC >>> 1037 NE 65th St Suite 273 >>> Seattle, WA 98115 >>> Main: (877) 211-9337 >>> Office: (206) 388-4772 >>> Cell: (253) 686-5518 >>> >>> william.king at quentustech.com >>> >>> >>> On 12/21/2012 03:54 AM, Antonio wrote: >>>> Answer to myself.... >>>> >>>> In the yealink configuration, in the account parameters, the "transport" >>>> must be force to TLS. >>>> >>>> I don't know why it just works.... Before i was using DNS-SRV, that >>>> should be the first option, yealink should have some issue here... i >>>> will report to them. >>>> >>>> >>>> Thanks, >>>> Ant?nio >>>> >>>> On Fri, 2012-12-21 at 10:35 +0100, Antonio wrote: >>>>> Hi, >>>>> >>>>> I'm trying to register a yealink with TLS, using my one certificates. >>>>> >>>>> I follow the wiki and In fs i have both agent.pem and cafile.pem . I >>>>> install in the phone the root certificate. >>>>> >>>>> But when i try to register, i have (tport log): >>>>> >>>>> >>>>> tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x808fb0) msg >>>>> 0x7fe9d0aa8180 from (udp/192.168.10.1:5060) has 340 bytes, veclen = 1 >>>>> tport.c:3004 tport_deliver() tport_deliver(0x808fb0): msg >>>>> 0x7fe9d0aa8180 (340 bytes) from udp/192.168.10.23:5060/sip next=(nil) >>>>> tport.c:4202 tport_release() tport_release(0x808fb0): 0x7fe9d01142f0 >>>>> by 0x7fe9d025d920 with 0x7fe9d0aa8180 >>>>> tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fe9c802aad0): >>>>> events IN >>>>> tport.c:869 tport_alloc_secondary() >>>>> tport_alloc_secondary(0x7fe9c802aad0): new secondary tport 0x7fe9c03e8450 >>>>> tport_type_tls.c:603 tport_tls_accept() >>>>> tport_tls_accept(0x7fe9c03e8450): new connection from >>>>> tls/192.168.10.36:48754/sips >>>>> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events >>>>> NEGOTIATING >>>>> tport_tls.c:869 tls_connect() tls_connect(0x7fe9c03e8450): events >>>>> NEGOTIATING >>>>> tport_tls.c:526 tls_post_connection_check() >>>>> tls_post_connection_check(0x7fe9c03e8450): Peer did not provide X.509 >>>>> Certificate. >>>>> >>>>> >>>>> >>>>> I could make it work and have a register in the tls profile when i >>>>> check on the phone the option in Security->Trusted Certificates: "Only >>>>> Accept Trusted Certificates: DISABLED". >>>>> Could it be some bug in the yealink, or I?m missing something in the >>>>> conf... >>>>> >>>>> Another question, is there any problem if i choose to use this >>>>> configuration... since is the phone that ignores the certificate and >>>>> the validation is done by the server and not by the client. >>>>> >>>>> Can you help me? >>>>> >>>>> Thanks, >>>>> Ant?nio >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> >>> consulting at freeswitch.org >>> >>>>> >>> http://www.freeswitchsolutions.com >>> >>>>> >>>>> >>>>> >>> >>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> >>> http://www.freeswitch.org >>> >>>>> >>> http://wiki.freeswitch.org >>> >>>>> >>> http://www.cluecon.com >>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>>>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>>> >>> http://www.freeswitch.org >>> >>>> >>>> -- >>>> >>>> Un cordial saludo / Best regards, >>>> >>>> _________________________ >>>> >>>> Ant?nio Silva >>>> >>>> E-mail: >>> asilva at wirelessmundi.com >>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> >>> consulting at freeswitch.org >>> >>>> >>> http://www.freeswitchsolutions.com >>> >>>> >>>> >>>> >>> >>> >>>> >>>> Official FreeSWITCH Sites >>>> >>> http://www.freeswitch.org >>> >>>> >>> http://wiki.freeswitch.org >>> >>>> >>> http://www.cluecon.com >>> >>>> >>>> FreeSWITCH-users mailing list >>>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>> >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From stephen at picardogroup.com Thu Jan 17 04:53:18 2013 From: stephen at picardogroup.com (stephen at picardogroup.com) Date: Wed, 16 Jan 2013 18:53:18 -0700 Subject: [Freeswitch-users] Freeswitch and lcr Message-ID: <20130116185318.0e1bd4d5c5064b420440751b21b10e46.fb7fcd6cd0.wbe@email13.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130116/225f3ef9/attachment-0001.html From =?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?= Thu Jan 17 07:05:37 2013 From: =?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?= (=?utf-8?Q?=D0=95=D1=80=D0=B6=D0=B0=D0=BD_=D0=A2=D1=83=D0=BB=D0=B5?=) Date: Thu, 17 Jan 2013 10:05:37 +0600 Subject: [Freeswitch-users] Unloop extension Message-ID: <20130117100537.56256009@mail.btcom.kz> FreeSWITCH 1.0.6 book on p. 187 says: ... When you set a variable during this phase, that variable is considered global automatically and becomes accessible throughout the application as $${variable} elsewhere in the XML. ... You can utilize global variables in your conditions, your variable and parameter declarations, pretty much anywhere, like this: This should had an addition, that ${g_v} can be used as well. ----- Original Message ----- From: Michael Collins [mailto:msc at freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Thu, 17 Jan 2013 07:17:31 +0600 Subject: Re: [Freeswitch-users] Unloop extension > No, it's available globally, whether used with $${} or ${} notation. You > can also see it from fs_cli: > > freeswitch at default> global_getvar unroll_loops > true > > More info about $${} vs. ${} is found > here > . > > -MC > > On Tue, Jan 15, 2013 at 8:09 PM, ????? ??????? > wrote: > > > Ain't what was set with > used in $${...} notation? > > > > ----- Original Message ----- > > From: Michael Collins [mailto:msc at freeswitch.org] > > To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > > Sent: Wed, 16 Jan 2013 06:54:12 +0600 > > Subject: Re: [Freeswitch-users] Unloop extension > > > > > > > Yes, it's okay. The pre-process does a set and ${unroll_loops} gets set > > as > > > needed. Then this dp entry does the work: > > > > > > > > > > > > > > > > > > > > > > > > More info on sip_looped_call var: > > > http://wiki.freeswitch.org/wiki/Variable_sip_looped_call > > > > > > -MC > > > > > > On Mon, Jan 14, 2013 at 9:33 PM, ????? ??????? > > > wrote: > > > > > > > What unloop extension is used for? BTW: vars.xml pre-(process)-sets > > > > unroll_loops, whereas DPs (public and default) are using channel > > variable > > > > syntax? Is this ok? > > > > > > > > Y. > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > -- > > > Michael S Collins > > > Twitter: @mercutioviz > > > http://www.FreeSWITCH.org > > > http://www.ClueCon.com > > > http://www.OSTAG.org > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > From hynek.cihlar at gmail.com Thu Jan 17 10:36:47 2013 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Thu, 17 Jan 2013 08:36:47 +0100 Subject: [Freeswitch-users] Early media handling on outbound call In-Reply-To: References: Message-ID: So it turns out to be problem on the provider's side. Sorry everybody for bothering. Hynek On Tue, Jan 15, 2013 at 12:57 PM, Steven Ayre wrote: > It seems odd that they immediately announce that it's not answering > then starts ringing. > > That's sounds like a problem with the VoIP operator - you'll need to > report it to them. > > Is the ringback tone being sent by the operator or generated locally > (eg by your softphone) when 180 Ringing is received? It would be > possible to set ignore_early_media=true which'd stop you seeing > anything until the 480. But in that case you'd also never hear any > ringback tone - unless you generated it on FS and in that case it > wouldn't necessarily mean the phone was actually ringing. > > -Steve > > > > On 15 January 2013 07:21, Hynek Cihlar wrote: > > Exactly. Here's the captured SIP flow: > > https://dl.dropbox.com/u/6276683/sip_flow.png. Between the time 4 and > 7, the > > mobile phone is hung up. After the the time 28 the mobile phone starts > > ringing again and hung up, the same on times 51 and 74. > > > > I am starting to think that this is something my VoIP operator should > deal > > with. What should be the expected flow? Status 480 right after the early > > media is played back? > > > > Hynek > > > > > > On Tue, Jan 15, 2013 at 12:18 AM, Michael Collins > > wrote: > >> > >> So you're saying that the provider is sending inband audio in early > media > >> saying that the call is not answering, and then on the same call you > hear > >> ringing? > >> > >> -MC > >> > >> On Sun, Jan 13, 2013 at 6:28 AM, Hynek Cihlar > >> wrote: > >>> > >>> Scenario: > >>> A call is originated to a mobile network. Mobile phone starts ringing > and > >>> is hung up. The mobile phone leg is pre-answered and early media is > returned > >>> saying that the caller is not answering. The message is playbed back > twice, > >>> than silence for cca two seconds and then ringing tone with the actual > phone > >>> starting to ring as well. The phone is hung up, early media played > back, > >>> etc... > >>> > >>> The freeswitch console command used to originate the call is as > follows: > >>> originate sofia/internal/1%terminal_domain > >>> &bridge({origination_caller_id_number= >>> number>}sofia/gateway/axfone/) > >>> > >>> My question is. Is this expected behavior? Should not the destination > >>> server end the call after early media is played back? Or is it > expected from > >>> me to detect a silence and end the call? If so, what would be the best > >>> approach to solve it, wait_for_silence? > >>> > >>> Thanks for comments! > >>> Hynek > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> > >> -- > >> Michael S Collins > >> Twitter: @mercutioviz > >> http://www.FreeSWITCH.org > >> http://www.ClueCon.com > >> http://www.OSTAG.org > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/bcd3f4f5/attachment-0001.html From hynek.cihlar at gmail.com Thu Jan 17 10:40:11 2013 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Thu, 17 Jan 2013 08:40:11 +0100 Subject: [Freeswitch-users] Escape characters in origination_caller_id_name Message-ID: Is there an escape character for the origination_caller_id_name variable so the full character set could be sent in the SIP packet? Hynek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/a57946d4/attachment.html From yehavi.bourvine at gmail.com Thu Jan 17 11:25:38 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Jan 2013 10:25:38 +0200 Subject: [Freeswitch-users] Softphone with BLF keys Message-ID: Hello, I am looking for a softphone (free or commercial) which supports true BLF. I mean: there are a few defined BLFs, which can be used to pickup the call when ringing. So far I could find nothing that really do the job. I think Bria does it, but once I did the mistake of importing Outlook contacts, the real ones I need are burried quite deeply in the list. Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/ac97fd07/attachment.html From avi at avimarcus.net Thu Jan 17 11:33:06 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 17 Jan 2013 10:33:06 +0200 Subject: [Freeswitch-users] Freeswitch and lcr In-Reply-To: <20130116185318.0e1bd4d5c5064b420440751b21b10e46.fb7fcd6cd0.wbe@email13.secureserver.net> References: <20130116185318.0e1bd4d5c5064b420440751b21b10e46.fb7fcd6cd0.wbe@email13.secureserver.net> Message-ID: /usr/local/src/freeswitch/modules.conf controls which mods get compiled. If you enable mod_lcr there, then when you build FS it will be included. What you want to modify is /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and make sure is in there. That controls which modules get loaded automatically when FS boots. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/14082fe4/attachment.html From gvvsubhashkumar at gmail.com Thu Jan 17 12:41:14 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Thu, 17 Jan 2013 15:11:14 +0530 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: Hi Michael, Below is the snippet of the freeswitch log. Success Call : 2013-01-16 00:58:58.314163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 2013-01-16 00:58:58.314163 [DEBUG] ftdm_io.c:3038 [s4c1][4:1] Created DTMF buffer 2013-01-16 00:58:58.314163 [DEBUG] switch_ivr_bridge.c:393 Send signal FreeTDM/4:1/2001 [BREAK] 2013-01-16 00:58:58.314163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating DTMF [9] 2013-01-16 00:59:06.414163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 1:640 2013-01-16 00:59:06.414163 [DEBUG] switch_ivr_bridge.c:393 Send signal FreeTDM/4:1/2001 [BREAK] 2013-01-16 00:59:06.414163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating DTMF [1] 2013-01-16 00:59:06.774163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 2013-01-16 00:59:06.774163 [DEBUG] switch_ivr_bridge.c:393 Send signal FreeTDM/4:1/2001 [BREAK] 2013-01-16 00:59:06.774163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating DTMF [9] 2013-01-16 00:59:09.214164 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read wanpipe event 3 2013-01-16 00:59:09.254162 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read wanpipe event 3 2013-01-16 00:59:09.254162 [DEBUG] ftmod_wanpipe.c:1547 [s4c1][4:1] Queuing wanpipe DTMF: 7 2013-01-16 00:59:09.254162 [DEBUG] ftdm_io.c:3783 [s4c1][4:1] Queuing DTMF 7 (debug = 0) 2013-01-16 00:59:09.254162 [DEBUG] mod_freetdm.c:811 Queuing DTMF [7] in channel FreeTDM/4:1/2001 device 4:1 2013-01-16 00:59:09.254162 [DEBUG] switch_ivr_bridge.c:393 Send signal sofia/VOIPBRD1/2001@**.**.**.** [BREAK] 2013-01-16 00:59:09.514163 [DEBUG] switch_rtp.c:2589 Queue digit delay of 40ms 2013-01-16 00:59:09.594164 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read wanpipe event 3 2013-01-16 00:59:09.654166 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read wanpipe event 3 2013-01-16 00:59:09.654166 [DEBUG] ftmod_wanpipe.c:1547 [s4c1][4:1] Queuing wanpipe DTMF: 1 2013-01-16 00:59:09.654166 [DEBUG] ftdm_io.c:3783 [s4c1][4:1] Queuing DTMF 1 (debug = 0) 2013-01-16 00:59:09.674165 [DEBUG] mod_freetdm.c:811 Queuing DTMF [1] in channel FreeTDM/4:1/2001 device 4:1 2013-01-16 00:59:09.674165 [DEBUG] switch_ivr_bridge.c:393 Send signal sofia/VOIPBRD1/2001@**.**.**.** [BREAK] 013-01-16 00:59:09.934163 [DEBUG] switch_rtp.c:2589 Queue digit delay of 40ms 2013-01-16 00:59:11.234163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 2013-01-16 00:59:11.234163 [DEBUG] switch_ivr_bridge.c:393 Send signal FreeTDM/4:1/2001 [BREAK] 2013-01-16 00:59:11.234163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating DTMF [9] 2013-01-16 00:59:11.394163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 2013-01-16 00:59:11.394163 [DEBUG] switch_ivr_bridge.c:393 Send signal FreeTDM/4:1/2001 [BREAK] 2013-01-16 00:59:11.394163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating DTMF [9] Failed DTMF call : 2013-01-16 01:02:27.694163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 2013-01-16 01:02:27.694163 [DEBUG] switch_ivr_bridge.c:393 Send signal FreeTDM/4:1/2001 [BREAK] 2013-01-16 01:02:27.694163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating DTMF [9] 2013-01-16 01:02:32.834164 [INFO] ftmod_sangoma_isdn_stack_rcv.c:232 [s4c1][4:1] Received DISCONNECT (suId:1 suInstId:2 spInstId:2) 2013-01-16 01:02:32.834164 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:543 [s4c1][4:1] Processing DISCONNECT (suId:1 suInstId:2 Thanks, Subhash. On Thu, Jan 17, 2013 at 6:48 AM, Michael Collins wrote: > Can you get a FS console log of this occurring? Also, what are the > endpoints that are being connected? > -MC > > On Wed, Jan 16, 2013 at 4:59 AM, Subhash wrote: > >> Hi all, >> >> I have a freeswitch configured with freetdm and wanpipe.My problem is >> with dtmf. When calling from an external pstn number to freeswitch, dtmf >> seems to be detected properly and enqueued it in the channel consistently. >> But when call came to same channel again, dtmf which is stored in the >> buffer for the previous call is used in the current call.So when i am done >> with previous call i want to dequeue the dtmf in the channel .So the >> problem seems to be something I've misconfigured with the Sangoma card. >> Can anyone point me in the right direction? >> >> Thanks, >> Subhash. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/e9fea601/attachment.html From muqtheear.s at gmail.com Thu Jan 17 13:00:17 2013 From: muqtheear.s at gmail.com (Muqtheear S) Date: Thu, 17 Jan 2013 15:30:17 +0530 Subject: [Freeswitch-users] Issue while playing MP4 video - Suggest me. Message-ID: Hi all, I am trying to play *minnale.mp4 *whose *mp4info* is as shown below. *# mp4info minnale.mp4* Track Type Info 1 video MPEG-4 Simple @ L1, 97.833 secs, 136 kbps, 176x144 @ 12.000041 fps 2 audio MPEG-4 AAC LC, 97.568 secs, 64 kbps, 32000 Hz 3 hint Payload MP4V-ES for track 1 4 hint Payload mpeg4-generic for track 2 My SIP client *Ekiga 4.0.0* which has *MP4V-ES* video codec support. I have enabled *MP4V-ES* video codec in Freeswitch, but there is no * mpeg4-generic* audio codec in both Ekiga client and Freeswitch server. I am streaming *minnale.mp4* from freeswitch server to *Ekiga* client. Now on my client side I could see only the video, but there is distortion in audio. How can I overcome above issue? Is there any alternate way to play *.mp4* files?. Please let me know. Thanks & Regards, Muqtheear.S -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/e73fe002/attachment.html From muqtheear.s at gmail.com Thu Jan 17 13:10:14 2013 From: muqtheear.s at gmail.com (Muqtheear S) Date: Thu, 17 Jan 2013 15:40:14 +0530 Subject: [Freeswitch-users] Issue - Recording and Playing MP4 video Message-ID: Hi all, My SIP client *Ekiga 4.0.0* which has *MP4V-ES* video codec support. I have enabled *MP4V-ES* video codec in Freeswitch. I am trying to record MP4 video as shown below. ** After recording, if I look at *mp4info* of *record_and_play.mp4, *it is showing as below #*mp4info /tmp/record_and_play.mp4* mp4info version 1.5.0.1 /tmp/record_and_play.mp4: mp4info: can't open /tmp/record_and_play.mp4 I am neither able to open in *VLC* player. Please tell me where I have gone wrong?. Thanks & Regards, Muqtheear.S -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/02c4cc44/attachment-0001.html From sertys at gmail.com Thu Jan 17 13:37:41 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 17 Jan 2013 11:37:41 +0100 Subject: [Freeswitch-users] Issue - Recording and Playing MP4 video In-Reply-To: References: Message-ID: Is the file at least there? Can u cat it? Can u post a verbose log of the record video call? On Jan 17, 2013 12:14 PM, "Muqtheear S" wrote: > Hi all, > My SIP client *Ekiga 4.0.0* which has *MP4V-ES* video codec > support. I have enabled *MP4V-ES* video codec in Freeswitch. > I am trying to record MP4 video as shown below. > ** > > After recording, if I look at *mp4info* of *record_and_play.mp4, *it is > showing as below > > #*mp4info /tmp/record_and_play.mp4* > mp4info version 1.5.0.1 > /tmp/record_and_play.mp4: > mp4info: can't open /tmp/record_and_play.mp4 > > I am neither able to open in *VLC* player. > Please tell me where I have gone wrong?. > > > Thanks & Regards, > Muqtheear.S > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/308eaa67/attachment.html From raimund.sacherer at logitravel.com Thu Jan 17 13:45:37 2013 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Thu, 17 Jan 2013 11:45:37 +0100 (CET) Subject: [Freeswitch-users] limit memory usage of freeswitch In-Reply-To: <32080636.8916.1358419529242.JavaMail.javamailuser@localhost> Message-ID: <17844111.8921.1358419535248.JavaMail.javamailuser@localhost> Hello Anthony, I also see Freeswitch eating away on Memory. Right now we are in the process of installing a new FS Cluster and I will upgrade all our PBX's with the new stable branch and I will test with that again. Our version of Freeswitch in production is: FreeSWITCH Version 1.0.head (git-4ceca18 2011-10-21 20-38-41 -0500) This version slowly grows and grows, Over month the biggest grow I saw was about 4-5 Gigabyte of RES memory. Before that we used a Freeswitch version from 2011-03 and that version grow to 273 Megabyte of memory usage and did not grow further. In the future I will report my findings with the new stable branch, hopefully whatever caused the memory consumption is allready fixed. Best regards, Ray ----- Original Message ----- From: "Anthony Minessale" To: "FreeSWITCH Users Help" Sent: Jueves, 17 de Enero 2013 2:12:58 Subject: Re: [Freeswitch-users] limit memory usage of freeswitch It does not eat memory.... if you attack it with sipp you may cause the internal pools to swell up but normal usage only consumes memory based on usage. On Wed, Jan 16, 2013 at 4:09 AM, jay binks < jaybinks at gmail.com > wrote: OpenVZ / Proxmox will allow you use set quota's and containerize your freeswitch... thats ONE way to do what your after. On 16 January 2013 19:48, rentmycoder rentmycoder < rentmycoder at gmail.com > wrote:
Hi All, According to my experineces and some info found on FS lists it is the normal behaviour that freeswitch eats up all of the available memory... It's not a memory leak, but the standard behaviour of FS by design... Correct me if I'm wrong... Is there any way to limit FS memory usage??? I need another services to run on the FS box... Thanks, John _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/21c13847/attachment.html From covici at ccs.covici.com Thu Jan 17 14:22:17 2013 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 17 Jan 2013 06:22:17 -0500 Subject: [Freeswitch-users] Mod_Conference Improvement (Denoise) In-Reply-To: <50F79C1F.8040804@quentustech.com> References: <5274F7F1-8BC9-4721-B98F-A163495CD730@jerris.com> <1358404199.68203.YahooMailNeo@web162901.mail.bf1.yahoo.com> <50F79C1F.8040804@quentustech.com> Message-ID: <17362.1358421737@ccs.covici.com> The problem with agc is thatit changed levels a lot within the same speaker and thus lessened the sound quality when I was using it. William King wrote: > That feature already exists, but appears to still be undocumented. Added > just over a year ago. > http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_23 > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 01/16/2013 10:29 PM, Valer Nur wrote: > > I think the best way to solve this is by doing AGC first. This should > > make sure audio level is similar for all participants. > > > > > > ------------------------------------------------------------------------ > > *From:* Anthony Minessale > > *To:* FreeSWITCH Users Help > > *Sent:* Wednesday, January 16, 2013 6:23 PM > > *Subject:* Re: [Freeswitch-users] Mod_Conference Improvement (Denoise) > > > > There is also the preprocess app which uses a media bug in the core for > > a variety of things but is incomplete it's using a bit of code from > > speex so we can certainly improve that as well. > > > > > > > > > > On Tue, Jan 15, 2013 at 8:58 PM, Michael Jerris > > wrote: > > > > If you wanted to implement something like this you could do so as a > > media bug, it does not need to touch mod_conference in any way. > > Take a look at mod_ladspa for an example of a similar module that > > modifies audio using a media bug. Give it a try and see if you can > > get something working. > > > > Mike > > > > On Jan 15, 2013, at 8:03 PM, Michael Collins > > wrote: > > > >> While I don't have a problem with the concept of noise filtering I > >> have to point out that each member of the conference can have his > >> or her own noise level setting. From fs_cli: > >> > >> conference energy > >> [] > >> > >> You only need to apply it to the person who is in the noisy > >> location. In fact, in the example configuration the user can dial > >> 9 to increase the energy threshold (for when he's in a noisy > >> environment) or press 7 to decrease the threshold. Pressing 8 will > >> reset the energy to normal, which I believe is 200. > >> > >> -MC > >> > >> On Tue, Jan 15, 2013 at 4:24 PM, Usama Zaidi >> > wrote: > >> > >> Hi, > >> > >> I talked about this a while back on the channel, the way how > >> energy-level is implemented in mod_conference isn't the right > >> way to handle noise in my opinion because the participants > >> that get bridged would have a different noise floor depending > >> on where they are located, so if I set the value of > >> energy-level too high to accommodate one participant, any > >> other participant in a quite sitting would never get bridged, > >> I proposed we should tackle this the way Asterisk does using a > >> denoise dialplan function > >> (https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE), > >> we can use libav for noise filtering on the muxed audio. I'm > >> willing to contribute if one of the devs would help me out > >> regarding implementation. Any comments? > >> > >> -Regards > >> > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > googletalk:conf+888 at conference.freeswitch.org > > > > pstn:+19193869900 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From itsusama at gmail.com Thu Jan 17 16:32:34 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Thu, 17 Jan 2013 18:32:34 +0500 Subject: [Freeswitch-users] Mod_Conference Improvement (Denoise) Message-ID: Hi, So I'm trying to borrow some code from sox now and trying to make an adaptive noise filter which would basically be equivalent of using the noiseprof and noisered filters on a stored file, I'll be calling the noiseprof function whenever there is silence on the line to update my noise profile after regular intervals lets see how that goes. On Thu, Jan 17, 2013 at 12:25 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Mod_Conference Improvement (Denoise) (Valer Nur) > 2. Re: Mod_Conference Improvement (Denoise) (William King) > 3. Re: 488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with > JsSIP and OverSIP + FreeSWITCH (royj) > > > ---------- Forwarded message ---------- > From: Valer Nur > To: FreeSWITCH Users Help > Cc: > Date: Wed, 16 Jan 2013 22:29:59 -0800 (PST) > Subject: Re: [Freeswitch-users] Mod_Conference Improvement (Denoise) > I think the best way to solve this is by doing AGC first. This should make > sure audio level is similar for all participants. > > > ------------------------------ > *From:* Anthony Minessale > *To:* FreeSWITCH Users Help > *Sent:* Wednesday, January 16, 2013 6:23 PM > *Subject:* Re: [Freeswitch-users] Mod_Conference Improvement (Denoise) > > There is also the preprocess app which uses a media bug in the core for a > variety of things but is incomplete it's using a bit of code from speex so > we can certainly improve that as well. > > > > > On Tue, Jan 15, 2013 at 8:58 PM, Michael Jerris wrote: > > If you wanted to implement something like this you could do so as a media > bug, it does not need to touch mod_conference in any way. Take a look at > mod_ladspa for an example of a similar module that modifies audio using a > media bug. Give it a try and see if you can get something working. > > Mike > > On Jan 15, 2013, at 8:03 PM, Michael Collins wrote: > > While I don't have a problem with the concept of noise filtering I have to > point out that each member of the conference can have his or her own noise > level setting. From fs_cli: > > conference energy [] > > You only need to apply it to the person who is in the noisy location. In > fact, in the example configuration the user can dial 9 to increase the > energy threshold (for when he's in a noisy environment) or press 7 to > decrease the threshold. Pressing 8 will reset the energy to normal, which I > believe is 200. > > -MC > > On Tue, Jan 15, 2013 at 4:24 PM, Usama Zaidi wrote: > > Hi, > > I talked about this a while back on the channel, the way how energy-level > is implemented in mod_conference isn't the right way to handle noise in my > opinion because the participants that get bridged would have a different > noise floor depending on where they are located, so if I set the value of > energy-level too high to accommodate one participant, any other participant > in a quite sitting would never get bridged, I proposed we should tackle > this the way Asterisk does using a denoise dialplan function ( > https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE), > we can use libav for noise filtering on the muxed audio. I'm willing to > contribute if one of the devs would help me out regarding implementation. > Any comments? > > -Regards > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: William King > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Wed, 16 Jan 2013 22:37:19 -0800 > Subject: Re: [Freeswitch-users] Mod_Conference Improvement (Denoise) > That feature already exists, but appears to still be undocumented. Added > just over a year ago. > http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_23 > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > On 01/16/2013 10:29 PM, Valer Nur wrote: > > I think the best way to solve this is by doing AGC first. This should > > make sure audio level is similar for all participants. > > > > > > ------------------------------------------------------------------------ > > *From:* Anthony Minessale > > *To:* FreeSWITCH Users Help > > *Sent:* Wednesday, January 16, 2013 6:23 PM > > *Subject:* Re: [Freeswitch-users] Mod_Conference Improvement (Denoise) > > > > There is also the preprocess app which uses a media bug in the core for > > a variety of things but is incomplete it's using a bit of code from > > speex so we can certainly improve that as well. > > > > > > > > > > On Tue, Jan 15, 2013 at 8:58 PM, Michael Jerris > > wrote: > > > > If you wanted to implement something like this you could do so as a > > media bug, it does not need to touch mod_conference in any way. > > Take a look at mod_ladspa for an example of a similar module that > > modifies audio using a media bug. Give it a try and see if you can > > get something working. > > > > Mike > > > > On Jan 15, 2013, at 8:03 PM, Michael Collins > > wrote: > > > >> While I don't have a problem with the concept of noise filtering I > >> have to point out that each member of the conference can have his > >> or her own noise level setting. From fs_cli: > >> > >> conference energy > >> [] > >> > >> You only need to apply it to the person who is in the noisy > >> location. In fact, in the example configuration the user can dial > >> 9 to increase the energy threshold (for when he's in a noisy > >> environment) or press 7 to decrease the threshold. Pressing 8 will > >> reset the energy to normal, which I believe is 200. > >> > >> -MC > >> > >> On Tue, Jan 15, 2013 at 4:24 PM, Usama Zaidi >> > wrote: > >> > >> Hi, > >> > >> I talked about this a while back on the channel, the way how > >> energy-level is implemented in mod_conference isn't the right > >> way to handle noise in my opinion because the participants > >> that get bridged would have a different noise floor depending > >> on where they are located, so if I set the value of > >> energy-level too high to accommodate one participant, any > >> other participant in a quite sitting would never get bridged, > >> I proposed we should tackle this the way Asterisk does using a > >> denoise dialplan function > >> ( > https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE), > >> we can use libav for noise filtering on the muxed audio. I'm > >> willing to contribute if one of the devs would help me out > >> regarding implementation. Any comments? > >> > >> -Regards > >> > >> > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > googletalk:conf+888 at conference.freeswitch.org > > > > pstn:+19193869900 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: royj > To: > Cc: > Date: Thu, 17 Jan 2013 11:24:25 +0400 > Subject: Re: [Freeswitch-users] 488 Not Acceptable Here > (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH > > http://jira.freeswitch.org/browse/FS-4606 > > On Thu, 17 Jan 2013 02:05:34 +0530 > Paan Singh wrote: > > > New SIP capture og with FreeSWITCH log syntax highlighting : > > http://pastebin.freeswitch.com/20457 > > > > > > On Thu, Jan 17, 2013 at 12:50 AM, Paan Singh >wrote: > > > > > Hi, > > > > > > I'm using JsSIP from a webpage to make a SIP call to FS, using OverSIP > as > > > a Websocket->SIP proxy. > > > > > > I'm able to Register successfully, but when I make a call from JsSIP > UA to > > > FreeSWITCH, I get a 180 Ringing, but after that I get a 488 Not > Acceptable > > > here.. Please help me figure out what I'm doing wrong here. > > > > > > > > > My inbound_codec_prefs is : > > > > > > > > > > > > Here is FreeSWITCH's log for the incoming call : > > > > > > ============================================= > > > port.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN > > > tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) > > > tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg > 0x2913be0 > > > from (udp/116.214.15.45:5080) has 3113 bytes, veclen = 1 > > > tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x2913be0 > (3113 > > > bytes) from udp/121.245.35.150:5080/sip next=(nil) > > > nta.c:2803 agent_recv_request() nta: received INVITE > > > sip:30201009 at 116.214.15.45 SIP/2.0 (CSeq 2141) > > > nta.c:3161 agent_aliases() nta: canonizing > sip:30201009 at 116.214.15.45with contact > > > nta.c:3002 agent_recv_request() nta: INVITE (2141) going to a default > leg > > > nta.c:1348 set_timeout() nta: timer set to 200 ms > > > nua_server.c:102 nua_stack_process_request() nua: > > > nua_stack_process_request: entering > > > nua_stack.c:899 nh_create() nua: nh_create: entering > > > nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering > > > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: > entering > > > soa.c:282 soa_clone() soa_clone(static::0x26122e0, 0x2606e20, > > > 0x7f91840c4c50) called > > > soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) > > > called > > > nta.c:4313 nta_leg_tcreate() nta_leg_tcreate(0x7f91840356d0) > > > soa.c:1302 soa_init_offer_answer() > > > soa_init_offer_answer(static::0x7f91840b25e0) called > > > soa.c:1171 soa_set_remote_sdp() > soa_set_remote_sdp(static::0x7f91840b25e0, > > > (nil), 0x285676c, 2333) called > > > nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f91840c4c50): adding > > > session usage > > > tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ > > > 121.245.35.150:5065 > > > tport.c:4026 tport_resolve() tport_resolve addrinfo = > 121.245.35.150:5065 > > > tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not > found > > > by name UDP/121.245.35.150:5065 > > > tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 564 bytes of 564 to > udp/ > > > 121.245.35.150:5065 > > > tport.c:3472 tport_send_msg() tport_vsend returned 564 > > > nta.c:6678 incoming_reply() nta: sent 100 Trying for INVITE (2141) > > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_invite > 100 > > > Trying > > > nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call > > > state changed: init -> received, received offer > > > soa.c:1098 soa_get_remote_sdp() > soa_get_remote_sdp(static::0x7f91840b25e0, > > > [0x7f91aa98a5e8], [0x7f91aa98a5e0], [(nil)]) called > > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state > 100 > > > Trying > > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > > entering > > > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > > > 2013-01-17 00:25:27.733506 [NOTICE] switch_channel.c:968 New Channel > > > sofia/external/1000 at 116.214.15.45:5080 > [4aec7fd8-600e-11e2-8706-1365ba01ad52] > > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send > signal > > > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > > entering > > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send > signal > > > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:415 > > > (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_NEW > > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:433 > > > (sofia/external/1000 at 116.214.15.45:5080) State NEW > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5576 Channel sofia/external/ > > > 1000 at 116.214.15.45:5080 entering state [received][100] > > > 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5587 Remote SDP: > > > v=0 > > > o=- 2957375511 2 IN IP4 127.0.0.1 > > > s=- > > > t=0 0 > > > a=group:BUNDLE audio video > > > m=audio 40111 RTP/SAVPF 103 104 0 8 106 105 13 126 > > > c=IN IP4 121.245.35.150 > > > a=rtcp:40111 IN IP4 121.245.35.150 > > > a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host > > > generation 0 > > > a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host > > > generation 0 > > > a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx > > > generation 0 > > > a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx > > > generation 0 > > > a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host > > > generation 0 > > > a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host > > > generation 0 > > > a=ice-ufrag:qiKF0ErbxaXldKI0 > > > a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw > > > a=ice-options:google-ice > > > a=mid:audio > > > a=rtcp-mux > > > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > > > inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 > > > a=rtpmap:103 ISAC/16000 > > > a=rtpmap:104 ISAC/32000 > > > a=rtpmap:0 PCMU/8000 > > > a=rtpmap:8 PCMA/8000 > > > a=rtpmap:106 CN/32000 > > > a=rtpmap:105 CN/16000 > > > a=rtpmap:13 CN/8000 > > > a=rtpmap:126 telephone-event/8000 > > > a=ssrc:4278252985 cname:fMHr4uVHPcpHFfGu > > > a=ssrc:4278252985 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx > > > a=ssrc:4278252985 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx00 > > > m=video 40111 RTP/SAVPF 100 101 102 > > > c=IN IP4 121.245.35.150 > > > a=rtcp:40111 IN IP4 121.245.35.150 > > > a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host > > > generation 0 > > > a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host > > > generation 0 > > > a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx > > > generation 0 > > > a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx > > > generation 0 > > > a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host > > > generation 0 > > > a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host > > > generation 0 > > > a=ice-ufrag:qiKF0ErbxaXldKI0 > > > a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw > > > a=ice-options:google-ice > > > a=mid:video > > > a=rtcp-mux > > > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > > > inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 > > > a=rtpmap:100 VP8/90000 > > > a=rtpmap:101 red/90000 > > > a=rtpmap:102 ulpfec/90000 > > > a=ssrc:169275022 cname:fMHr4uVHPcpHFfGu > > > a=ssrc:169275022 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx > > > a=ssrc:169275022 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx10 > > > > > > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 > > > AES_CM_128_HMAC_SHA1_80 > inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2] > > > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:3155 Set Local Key [1 > > > AES_CM_128_HMAC_SHA1_80 > inline:i8vqmBeU4+/sjJBndCbEQ4G87gmpYeVnqxh8Xuzu] > > > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:5277 No 2833 in SDP. > > > Disable 2833 dtmf and switch to INFO > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:2994 > > > (sofia/external/1000 at 116.214.15:45:5080) Callstate Change DOWN -> > HANGUP > > > 2013-01-17 00:25:27.753511 [NOTICE] sofia.c:5878 Hangup sofia/external/ > > > 1000 at 116.214.15.45:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION] > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:3017 Send signal > > > sofia/external/1000 at 116.214.15.45:5080 [KILL] > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send > signal > > > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 > > > (sofia/external/1000 at 116.214.15.45:5080) Running State Change > CS_HANGUP > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 > > > (sofia/external/1000 at 116.214.15.45:5080) State HANGUP > > > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:503 Channel > sofia/external/ > > > 1000 at 116.214.15.45:5080 hanging up, cause: INCOMPATIBLE_DESTINATION > > > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:633 Responding to INVITE > > > with: 488 > > > nua.c:879 nua_respond() nua: nua_respond: entering > > > nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_respond > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:48 > > > sofia/external/1000 at 116.214.15.45:5080 Standard HANGUP, cause: > > > INCOMPATIBLE_DESTINATION > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 > > > (sofia/external/1000 at 116.214.15.45:5080) State HANGUP going to sleep > > > nua_stack.c:573 nua_stack_signal() nua(0x7f91840c4c50): recv signal > > > r_respond 488 Not Acceptable Here > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:446 > > > (sofia/external/1000 at 116.214.15.45:5080) State Change CS_HANGUP -> > > > CS_REPORTING > > > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: > entering > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send > signal > > > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > > > soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) > > > called > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 > > > (sofia/external/1000 at 116.214.15.45:5080) Running State Change > CS_REPORTING > > > nua_session.c:2316 nua_invite_server_respond() nua: > > > nua_invite_server_respond: entering > > > soa.c:1214 soa_clear_remote_sdp() > > > soa_clear_remote_sdp(static::0x7f91840b25e0) called > > > tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ > > > 121.245.35.150:5065 > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 > > > (sofia/external/1000 at 116.214.15.45:5080) State REPORTING > > > tport.c:4026 tport_resolve() tport_resolve addrinfo = > 121.245.35.150:5065 > > > tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not > found > > > by name UDP/121.245.35.150:5065 > > > tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 784 bytes of 784 to > udp/ > > > 121.245.35.150:5065 > > > tport.c:3472 tport_send_msg() tport_vsend returned 784 > > > nta.c:6678 incoming_reply() nta: sent 488 Not Acceptable Here for > INVITE > > > (2141) > > > nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f91840c4c50): > > > removing session usage > > > nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call > > > state changed: received -> terminated > > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state > 488 > > > Not Acceptable Here > > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event > i_terminated > > > 488 Not Acceptable Here > > > soa.c:358 soa_destroy() soa_destroy(static::0x7f91840b25e0) called > > > nta.c:4366 nta_leg_destroy() nta_leg_destroy(0x7f91840356d0) > > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > > entering > > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > > entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering > > > nua_stack.c:569 nua_stack_signal() nua(0x7f91840c4c50): recv signal > > > r_destroy > > > nta.c:4366 nta_leg_destroy() nta_leg_destroy((nil)) > > > nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_destroy > > > 2013-01-17 00:25:27.753511 [DEBUG] mod_cdr_sqlite.c:102 Writing SQL to > DB: > > > INSERT INTO cdr VALUES ("1000","1000","30201009","public","2013-01-17 > > > 00:25:27","","2013-01-17 > > > > 00:25:27",0,0,"INCOMPATIBLE_DESTINATION","4aec7fd8-600e-11e2-8706-1365ba01ad52","","") > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:92 > > > sofia/external/1000 at 116.214.15.45:5080 Standard REPORTING, cause: > > > INCOMPATIBLE_DESTINATION > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 > > > (sofia/external/1000 at 116.214.15.45:5080) State REPORTING going to > sleep > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:440 > > > (sofia/external/1000 at 116.214.15.45:5080) State Change CS_REPORTING -> > > > CS_DESTROY > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send > signal > > > sofia/external/1000 at 116.214.15.45:5080 [BREAK] > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1499 Session > 1794 > > > (sofia/external/1000 at 116.214.15.45:5080) Locked, Waiting on external > > > entities > > > 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1517 Session > > > 1794 (sofia/external/1000 at 116.214.15.45:5080) Ended > > > 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1521 Close > > > Channel sofia/external/1000 at 116.214.15.45:5080 [CS_DESTROY] > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:556 > > > (sofia/external/1000 at 116.214.15.45:5080) Callstate Change HANGUP -> > DOWN > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:559 > > > (sofia/external/1000 at 116.214.15.45:5080) Running State Change > CS_DESTROY > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 > > > (sofia/external/1000 at 116.214.15.45:5080) State DESTROY > > > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:396 sofia/external/ > > > 1000 at 116.214.15.45:5080 SOFIA DESTROY > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:99 > > > sofia/external/1000 at 116.214.15.45:5080 Standard DESTROY > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 > > > (sofia/external/1000 at 116.214.15.45:5080) State DESTROY going to sleep > > > nta.c:1294 agent_timer() nta: timer set next to 322 ms > > > tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN > > > tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) > > > tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg > > > 0x7f9198220070 from (udp/116.214.15.45:5080) has 339 bytes, veclen = 1 > > > tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg > 0x7f9198220070 > > > (339 bytes) from udp/121.245.35.150:5080/sip next=(nil) > > > nta.c:2803 agent_recv_request() nta: received ACK > > > sip:30201009 at 116.214.15.45 SIP/2.0 (CSeq 2141) > > > nta.c:2938 agent_recv_request() nta: ACK (2141) is going to INVITE > (2141) > > > nta.c:1294 agent_timer() nta: timer set next to 4823 ms > > > nta.c:7010 _nta_incoming_timer() nta: timer I fired, terminate 488 > response > > > nta.c:5718 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), > > > 0x7f91aa98ac20) > > > nta.c:7061 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 > tout, > > > 1/1 term, 1/1 free > > > nta.c:1279 agent_timer() nta: timer not set > > > ================================================================ > > > > > > Here is the Log from JsSIP in the browser that receives messages from > > > FreeSWITCH : > > > =========================================================== > > > JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 180 > Ringing > > > Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 Record-Route: > > > Record-Route: > > > > From: < > > > sip:1000 at 116.214.15.45:5080>;tag=rj2nc38qem To: < > > > sip:30201009 at 116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: > > > joshempnx2kuh7smrt3x CSeq: 1974 INVITE Contact: > > > User-Agent: > > > FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Accept: > > > application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, > INFO, > > > UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, > > > replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 > > > P-Asserted-Identity: "30201009" > > > jssip-0.2.1.js:684< > http://example.com:8080/static/did/tryit/jssip-0.2.1.js> > > > JsSIP | DIALOG | New UAC dialog created: 1 jssip-0.2.1.js:2171< > http://example.com:8080/static/did/tryit/jssip-0.2.1.js> > > > JsSIP | EVENT EMITTER | Emitting event: progress jssip-0.2.1.js:170< > http://example.com:8080/static/did/tryit/jssip-0.2.1.js> > > > JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 488 Not > > > Acceptable Here Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 > > > Max-Forwards: 10 From: ;tag=rj2nc38qem > To: < > > > sip:30201009 at 116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: > > > joshempnx2kuh7smrt3x CSeq: 1974 INVITE User-Agent: > > > FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Allow: > INVITE, > > > ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, > NOTIFY > > > Supported: timer, precondition, path, replaces Allow-Events: talk, > hold, > > > conference, refer Reason: > Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > > > Content-Length: 0 P-Asserted-Identity: "30201009" < > > > sip:30201009 at 116.214.15.45> > > > > ============================================================================= > > > > > > > > > > > > > > > > > > > > > What am I doing wrong? Thanks, > > > Paan > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/8da09dad/attachment-0001.html From yehavi.bourvine at gmail.com Thu Jan 17 16:37:04 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Jan 2013 15:37:04 +0200 Subject: [Freeswitch-users] Polycom with TLS? Message-ID: Hello, I have a long standing issue with Polycom and Freeswitch: The TLS session starts ok, I can make 1-2 calls, but then the TLS session dies. It re-connect automatically, but once ebwery fwe minuts it disconnects. Has anyone managed to make this working successfully? Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/274d76a5/attachment.html From frank at carmickle.com Thu Jan 17 16:38:43 2013 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 17 Jan 2013 08:38:43 -0500 Subject: [Freeswitch-users] pre_answer and dtmf Message-ID: <1FDA6E69-F4E0-4FDF-A5C9-8E86B5D94772@carmickle.com> Hello I can't seem to read digits with pre_answer. Answer works just fine. I know this used to work. I'm not sure if it's upstreams problem or mine. Where should I look? Thanks --FC From haloha201 at gmail.com Thu Jan 17 17:06:34 2013 From: haloha201 at gmail.com (haloha) Date: Thu, 17 Jan 2013 21:06:34 +0700 Subject: [Freeswitch-users] need help on call recording - Solved Message-ID: hi Benoit Raymond it works properly now :) Thank you for your guide Ha On 1/16/13, Benoit Raymond wrote: > Hi, > > It is most likely recorded but since you have RECORD_STEREO=true, the audio > is ... stereo! > > If you load your wav in an editor (like Goldwave), that should be evident. > So drop the RECORD_STEREO parameter or set it to false (this is the default > anyway) and check if that works. > > Benoit Raymond > AMI Concept Inc. > Business VoIP Solutions / Web Design > Phone: (450) 553-1231 > http://www.amiconcept.com > > -----Message d'origine----- > De : freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de haloha > Envoy? : 16 janvier 2013 03:50 > ? : FreeSWITCH Users Help > Objet : [Freeswitch-users] need help on call recording > > hi all > > i try to record all calls to/from local extension so my dialplan > > > > > > > > > > > data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> > > data="/etc/freeswitch/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${caller_id_ > number}_${destination_number}.wav"/> > > > > > > > > > > > > > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb > er}"/> > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/> > data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}" > /> > data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > data="user/${dialed_extension}@${domain_name}"/> > > > data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> > > > > > FS only records the caller voicem FS doesnt record the called voice > > How do i record the entire calls(caller and called) > > Thank you > Ha` > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From paul at cupis.co.uk Thu Jan 17 17:11:40 2013 From: paul at cupis.co.uk (Paul Cupis) Date: Thu, 17 Jan 2013 14:11:40 +0000 Subject: [Freeswitch-users] pre_answer and dtmf In-Reply-To: <1FDA6E69-F4E0-4FDF-A5C9-8E86B5D94772@carmickle.com> References: <1FDA6E69-F4E0-4FDF-A5C9-8E86B5D94772@carmickle.com> Message-ID: <20130117141140.GA5548@eagle.cupis.co.uk> On Thu, Jan 17, 2013 at 08:38:43AM -0500, Frank Carmickle wrote: > I can't seem to read digits with pre_answer. Answer works just fine. I know this used to work. I'm not sure if it's upstreams problem or mine. Where should I look? pre_answer will use Early Media, which will be one-way audio, so I would not expect you to be able to read digits from the caller unless you answer the call. Regards, From frank at carmickle.com Thu Jan 17 17:56:16 2013 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 17 Jan 2013 09:56:16 -0500 Subject: [Freeswitch-users] pre_answer and dtmf and now intercept_unanswered_only In-Reply-To: <20130117141140.GA5548@eagle.cupis.co.uk> References: <1FDA6E69-F4E0-4FDF-A5C9-8E86B5D94772@carmickle.com> <20130117141140.GA5548@eagle.cupis.co.uk> Message-ID: <3C31F543-6EAB-49F1-861F-EC5D289B74BD@carmickle.com> On Jan 17, 2013, at 9:11 AM, Paul Cupis wrote: > On Thu, Jan 17, 2013 at 08:38:43AM -0500, Frank Carmickle wrote: >> I can't seem to read digits with pre_answer. Answer works just fine. I know this used to work. I'm not sure if it's upstreams problem or mine. Where should I look? > > pre_answer will use Early Media, which will be one-way audio, so > I would not expect you to be able to read digits from the caller > unless you answer the call. > I'm trying to use intercept_unanswered_only=treu so as not to allow people to intercept a call that goes to voicemail. Of course this is after coming out of an IVR. Recommendations? Thanks --FC From jmesquita at freeswitch.org Thu Jan 17 18:05:06 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 17 Jan 2013 12:05:06 -0300 Subject: [Freeswitch-users] FAX t38_gateway operation confusion (mod_spandsp) Message-ID: Hello everyone! I have been playing around with mod_spandsp's capabilities lately and most certainly because I am new to the all old faxing stuff (pretty funny, right?), I have been making a bit of a mess in my head about how a T30->T38 gateway should work. I've set up a testing environment here as follows: *Components:* * * 2 analog fax machines (call is FaxA and FaxB) 2 FXS cards 2 FreeSWITCH machines (call FSA and FSB) *Connection scheme:* * * *FaxA <-> FSA and FaxB <-> FSB* Both of these connections are using the respective FXS cards *FSA <-SIP-> FSB* * * Both FreeSWITCHes are connected using SIP, of course. *Call scenarios:* 1. *FaxA sending a fax to FSB (using rxfax on FSB) - OK* Dialplan on FSA is: Dialplan on FSB: 2. *FSB sending a a fax to FaxA - OK* Dialplan on FSA is: Originate line on FSB: originate {fax_enable_t38_request=true,fax_enable_t38=true}sofia/internal/FSB 'set:ignore_early_media=true,txfax:/root/fax.tiff' inline 3. *FaxA sending a fax to FaxB in manual mode* 4. *FaxA sending a fax to FaxB in automatic mode* 5. *FaxA receiving a fax from FaxB in manual mode* 6. *FaxA receiving a fax from FaxB in automatic mode* Here comes the problem. It is not clear to me how this should work from the dialplan perspective and from the theory as well. Now, as I understand, t38_gateway works by detecting CED tones as specified on the code by the line: *spandsp_fax_detect_session(session, "rw", timeout, MODEM_CONNECT_TONES_FAX_CED_OR_PREAMBLE, 1, direction, NULL, t38_gateway_start);* *Questions:* If this is true and looking at the T.30 standard, some of the cases mentioned above make the CED tone optional, is this correct? By talking to Steve Underwood yesterday on IRC, he mentioned that (and I quote): "the emitting gateway should detect the signal from the answering FAX machine and initiate the change to T.38". My understanding is that on 3 for example, FSA needs to detect CED being sent from FaxB, correct? Looking at the T.38 standard Annex D to understand who should initiate a T.38 re-invite at what instance, I found this phrase: > "The receiving gateway should only use the ITU-T V.21 preamble generated > by the answering FAX endpoint as the trigger for detection, and it should > generate the INVITE request no more than 200 milliseconds after the ITU-T > V.21 preamble begins, in order to allow the maximum amount of time for the > session to transition into ITU-T T.38 mode before the answering FAX > endpoint begins transmitting NSF, CSI or DIS." Googling, I found that the ITU-T V.21 preamble is not CNG or CED tone, but rather something else... I assume that the _OR_PREAMBLE is exactly that, correct? I greatly appreciate any help as the Wiki is not very specific on the analog fax stuff. My intent is to wikify all that and help the folks out there as well. Fax seems to be mistic for more than myself. Regards, Jo?o Mesquita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/663dfd6d/attachment.html From jnvines at gmail.com Thu Jan 17 18:08:58 2013 From: jnvines at gmail.com (Nick Vines) Date: Thu, 17 Jan 2013 10:08:58 -0500 Subject: [Freeswitch-users] SIP Message Parsing and Auto-Action Message-ID: I have a couple Grandstream HT502s that occasionally get stuck with the detected NAT type of Warning: 399 ip.address "Detected NAT type is UDP Blocked". When that happens the device is never able to complete a call until it is restarted. Where should I start looking to create a monitor that will parse those messages and send a reboot (i.e. sofia profile [profile] flush_inbound_reg [call_id] reboot) Thanks, Nick *Sample SIP Message* SIP/2.0 200 OK Via: SIP/2.0/TCP XXX.XXX.XXX.XXX:PORT;branch=z9hG4bK2N9vy7BmyF40B From: ;tag=3yNZa6ZNv6K6g To: ;tag=1662499825 Call-ID: 3901be5e-db58-1230-109f-bc764e103a41 CSeq: 38881081 NOTIFY Supported: replaces, path, timer, eventlist User-Agent: Grandstream HT-502 V1.2A 1.0.8.4 chip V2.2 Warning: 399 XXX.XXX.XXX.XXX *"Detected NAT type is UDP Blocked"* Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/3f5338ad/attachment-0001.html From steveayre at gmail.com Thu Jan 17 19:52:24 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 17 Jan 2013 16:52:24 +0000 Subject: [Freeswitch-users] pre_answer and dtmf In-Reply-To: <20130117141140.GA5548@eagle.cupis.co.uk> References: <1FDA6E69-F4E0-4FDF-A5C9-8E86B5D94772@carmickle.com> <20130117141140.GA5548@eagle.cupis.co.uk> Message-ID: Yep, most phones won't send RTP which'll include DTMF until 200 OK (answer) is received... That will depend on the caller really so perhaps it does work on certain phones/versions, but not all will allow it. -Steve On 17 January 2013 14:11, Paul Cupis wrote: > On Thu, Jan 17, 2013 at 08:38:43AM -0500, Frank Carmickle wrote: > > I can't seem to read digits with pre_answer. Answer works just fine. I > know this used to work. I'm not sure if it's upstreams problem or mine. > Where should I look? > > pre_answer will use Early Media, which will be one-way audio, so > I would not expect you to be able to read digits from the caller > unless you answer the call. > > Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/123e64bb/attachment.html From moises.silva at gmail.com Thu Jan 17 19:55:56 2013 From: moises.silva at gmail.com (Moises Silva) Date: Thu, 17 Jan 2013 11:55:56 -0500 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: On Thu, Jan 17, 2013 at 4:41 AM, Subhash wrote: > Hi Michael, > > Below is the snippet of the freeswitch log. > > Success Call : > > 2013-01-16 00:58:58.314163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 > 2013-01-16 00:58:58.314163 [DEBUG] ftdm_io.c:3038 [s4c1][4:1] Created DTMF > buffer > 2013-01-16 00:58:58.314163 [DEBUG] switch_ivr_bridge.c:393 Send signal > FreeTDM/4:1/2001 [BREAK] > 2013-01-16 00:58:58.314163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating > DTMF [9] > 2013-01-16 00:59:06.414163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 1:640 > 2013-01-16 00:59:06.414163 [DEBUG] switch_ivr_bridge.c:393 Send signal > FreeTDM/4:1/2001 [BREAK] > 2013-01-16 00:59:06.414163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating > DTMF [1] > 2013-01-16 00:59:06.774163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 > 2013-01-16 00:59:06.774163 [DEBUG] switch_ivr_bridge.c:393 Send signal > FreeTDM/4:1/2001 [BREAK] > 2013-01-16 00:59:06.774163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating > DTMF [9] > 2013-01-16 00:59:09.214164 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read > wanpipe event 3 > 2013-01-16 00:59:09.254162 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read > wanpipe event 3 > 2013-01-16 00:59:09.254162 [DEBUG] ftmod_wanpipe.c:1547 [s4c1][4:1] > Queuing wanpipe DTMF: 7 > 2013-01-16 00:59:09.254162 [DEBUG] ftdm_io.c:3783 [s4c1][4:1] Queuing DTMF > 7 (debug = 0) > 2013-01-16 00:59:09.254162 [DEBUG] mod_freetdm.c:811 Queuing DTMF [7] in > channel FreeTDM/4:1/2001 device 4:1 > 2013-01-16 00:59:09.254162 [DEBUG] switch_ivr_bridge.c:393 Send signal > sofia/VOIPBRD1/2001@**.**.**.** [BREAK] > 2013-01-16 00:59:09.514163 [DEBUG] switch_rtp.c:2589 Queue digit delay of > 40ms > 2013-01-16 00:59:09.594164 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read > wanpipe event 3 > 2013-01-16 00:59:09.654166 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read > wanpipe event 3 > 2013-01-16 00:59:09.654166 [DEBUG] ftmod_wanpipe.c:1547 [s4c1][4:1] > Queuing wanpipe DTMF: 1 > 2013-01-16 00:59:09.654166 [DEBUG] ftdm_io.c:3783 [s4c1][4:1] Queuing DTMF > 1 (debug = 0) > 2013-01-16 00:59:09.674165 [DEBUG] mod_freetdm.c:811 Queuing DTMF [1] in > channel FreeTDM/4:1/2001 device 4:1 > 2013-01-16 00:59:09.674165 [DEBUG] switch_ivr_bridge.c:393 Send signal > sofia/VOIPBRD1/2001@**.**.**.** [BREAK] > 013-01-16 00:59:09.934163 [DEBUG] switch_rtp.c:2589 Queue digit delay of > 40ms > 2013-01-16 00:59:11.234163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 > 2013-01-16 00:59:11.234163 [DEBUG] switch_ivr_bridge.c:393 Send signal > FreeTDM/4:1/2001 [BREAK] > 2013-01-16 00:59:11.234163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating > DTMF [9] > 2013-01-16 00:59:11.394163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 > 2013-01-16 00:59:11.394163 [DEBUG] switch_ivr_bridge.c:393 Send signal > FreeTDM/4:1/2001 [BREAK] > 2013-01-16 00:59:11.394163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating > DTMF [9] > Failed DTMF call : > > 2013-01-16 01:02:27.694163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 > 2013-01-16 01:02:27.694163 [DEBUG] switch_ivr_bridge.c:393 Send signal > FreeTDM/4:1/2001 [BREAK] > 2013-01-16 01:02:27.694163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating > DTMF [9] > 2013-01-16 01:02:32.834164 [INFO] ftmod_sangoma_isdn_stack_rcv.c:232 > [s4c1][4:1] Received DISCONNECT (suId:1 suInstId:2 spInstId:2) > 2013-01-16 01:02:32.834164 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:543 > [s4c1][4:1] Processing DISCONNECT (suId:1 suInstId:2 > > This looks like incoming DTMF from the RTP leg going out in the TDM leg? if so, the problem would be why switch_rtp.c is "detecting" received DTMF when there is none? Please post in pastebin the full debug log (from call start, bridge, to hangup etc) for one success call followed by the call with the dtmf problem. *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/53bd7e27/attachment.html From akostenko at broadvox.com Thu Jan 17 19:39:14 2013 From: akostenko at broadvox.com (Alex Kostenko) Date: Thu, 17 Jan 2013 18:39:14 +0200 Subject: [Freeswitch-users] Freeswitch as SBC and refer method Message-ID: <50F82932.6030504@broadvox.com> Hi all We have next situation: We use Freeswitch as SBC. And now we stumbled to the situation when we need to translate "refer method" through freeswitch. Now freeswitch processes this option. This question appeared because of wrong amount of session from customer (more then limit for this customer). In some way customer can create more sessions using "refer method" more then he can. From ashou at mobile-sphere.com Thu Jan 17 20:02:51 2013 From: ashou at mobile-sphere.com (ashou) Date: Thu, 17 Jan 2013 09:02:51 -0800 (PST) Subject: [Freeswitch-users] Process REFER and not Hangup the channel Message-ID: <1358442171015-7586441.post@n2.nabble.com> Hi There, I use FS as proxy to handle a call with REFER to. My question is when get a REFER request, how to do the Transfer regarding the REFER TO before hangup the current channel. My mean before send back the NOTIFY, do the INVITE (based on REFER TO), when the INVITE is 200 OK, then send back NOTIFY. my logs: 11:24:14.141700 [DEBUG] sofia.c:6616 Process REFER to [11004 at 111.22.33.1] 11:24:14.141700 [DEBUG] switch_ivr.c:1773 (sofia/external/6175552222 at 12.13.14.15) State Change CS_EXECUTE -> CS_ROUTING 11:24:14.141700 [DEBUG] switch_core_session.c:1287 Send signal sofia/external/6175552222 at 12.13.14.15 [BREAK] 11:24:14.141700 [DEBUG] switch_core_session.c:830 Send signal sofia/external/6175552222 at 12.13.14.15 [BREAK] 11:24:14.141700 [NOTICE] switch_ivr.c:1779 Transfer sofia/external/6175552222 at 12.13.14.15 to XML[11004 at public] 11:24:14.161701 [DEBUG] switch_ivr_play_say.c:1682 done playing file local_stream://moh 11:24:14.161701 [DEBUG] switch_ivr_bridge.c:597 BRIDGE THREAD DONE [sofia/external/+18550000000 at 1.2.3.4:5060] 11:24:14.161701 [DEBUG] switch_ivr_bridge.c:622 Send signal sofia/external/6175552222 at 12.13.14.15 [BREAK] 11:24:14.161701 [DEBUG] switch_channel.c:2979 (sofia/external/+18550000000 at 1.2.3.4:5060) Callstate Change HELD -> HANGUP the Transfer to XML[11004 at public] not DONE yet, the (sofia/external/+18550000000 at 1.2.3.4:5060) already Hangup, How to delay the Hangup after Transfer is done? Thanks, Ashou -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Process-REFER-and-not-Hangup-the-channel-tp7586441.html Sent from the freeswitch-users mailing list archive at Nabble.com. From admin at smallunix.net Thu Jan 17 21:46:05 2013 From: admin at smallunix.net (Andrea Mazzeo) Date: Thu, 17 Jan 2013 19:46:05 +0100 Subject: [Freeswitch-users] SONUS - dtg inside INVITE Message-ID: Hi all, I need to configure my FS to send call to a Sonus GSX via SIP. Is there a way to send SIP parameter "dtg" (destination trunk group) inside the INVITE? Example: Request-Line: INVITE sip: 39340xxxxxx at 38.xxx.xxx.xxx SIP/2.0 dtg=ATTMDT7807IPO Thanks, Andrea From kris at kriskinc.com Thu Jan 17 23:58:08 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 17 Jan 2013 15:58:08 -0500 Subject: [Freeswitch-users] SONUS - dtg inside INVITE In-Reply-To: References: Message-ID: This is almost certainly not the syntax you want to use. Do that expect "dtg" to be a user or domain parameter? On Thu, Jan 17, 2013 at 1:46 PM, Andrea Mazzeo wrote: > Hi all, > > I need to configure my FS to send call to a Sonus GSX via SIP. > Is there a way to send SIP parameter "dtg" (destination trunk group) > inside the INVITE? > > Example: > > Request-Line: INVITE sip: 39340xxxxxx at 38.xxx.xxx.xxx SIP/2.0 dtg=ATTMDT7807IPO > > > Thanks, > Andrea > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From gamar at center.com Thu Jan 17 23:54:14 2013 From: gamar at center.com (Gilbert Amar) Date: Thu, 17 Jan 2013 12:54:14 -0800 Subject: [Freeswitch-users] building mod_opal and v1.2stable Message-ID: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> Hi, I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and opal 3.10.7 on a debian Ptlib and opal get build with no error, but when trying mod_opal I get Compiling /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DP_64BIT -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, switch_frame_t**, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: error: 'class OpalCall' has no member named 'IsSwitchingT38' /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, const switch_frame_t*, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: error: 'class OpalCall' has no member named 'IsSwitchingT38' make[3]: *** [mod_opal.lo] Error 1 make[2]: *** [install] Error 1 make[1]: *** [mod_opal-install] Error 1 make: *** [mod_opal-install] Error 2 Has anyone succeded in building mod_opal ? Should I go with mod_h323 instead ? Thanks for your help From jtbock at synacktics.com Fri Jan 18 01:15:36 2013 From: jtbock at synacktics.com (Tim Bock) Date: Thu, 17 Jan 2013 15:15:36 -0700 Subject: [Freeswitch-users] dtmf problem In-Reply-To: <50F5B089.3060000@synacktics.com> References: <50F38BEE.6090804@synacktics.com> <50F57D9F.4050000@tangramltd.com> <50F5B089.3060000@synacktics.com> Message-ID: <50F87808.9040807@synacktics.com> Hi, Thought I would update this for list posterity. Although the spandsp_start_dtmf trick seemed to work for me, I soon discovered that I now had an echo problem, previously banished with oslec. Indeed, lsmod still showed that oslec was loaded, and dahdi_cfg claimed that it was enabled for the channels in question. But when I would make or receive calls through the tdm card, I would not see the output of "cat /proc/dahdi/1" change. It remained "OSLEC - INACTIVE." I also began poring through the logs (past and present), and soon realized that freetdm does start a software dtmf detector (and in fact this had worked in the past). So, there was something else going on. I was trying various things, and looking through the list and docs, when (seemingly) all of a sudden, the echo went away and now the dtmf was detecting two keypresses for every single keypress. Ok, sure, understandable since I had turned on spandsp_dtmf in addition to freetdm's dtmf detector, but why now? And why did the echo magically disappear? Eventually, I figured out that it was related to me connecting/disconnecting the fxo port, which I do since I'm using my home line for testing. Something (freetdm or dahdi, presumably) doesn't like that. What fixes it is to have the trunk line connected when the freeswitch server is rebooted. There is undoubtedly a "smaller hammer" solution, but I haven't figured it out yet. Just stopping and restarting freeswitch doesn't do it. The upshot is that I removed the spandsp_start_dtmf from my dialplan. Rebooting with the trunk port occupied fixed all of the sins. Tim On 01/15/2013 12:39 PM, Tim Bock wrote: > I can tell you how I solved the problem I was having. Don't know if > this is preferred, or if there is a better way, but I'm now able to > detect dtmf from all sources (pstn, cell, sip). > > Apparently, mod spandsp has better dtmf detection than the dtmf > detection in the freeswitch core. On the wiki page for spandsp > (http://wiki.freeswitch.org/wiki/Mod_spandsp), there is a section on > dtmf. I added the spandsp_start_dtmf parameter before calling the > ivr, and that fixed my issue. > > Hope this helps, > Tim > > On 01/15/2013 09:02 AM, Viacheslav Dubrovskyi wrote: >> 14.01.2013 06:39, Tim Bock ?????: >>> I have a freeswitch server with a digium tdm wildcard. Compiled, >>> enabled, and configured freetdm and dahdi...all seems to be working fine >>> (can make/receive calls). >>> >>> My problem is with dtmf. When calling from an external pstn number to >>> freeswitch, dtmf seems to be detected properly and consistently. But >>> when calling from a cell phone or via another voip line, dtmf doesn't >>> seem to be detected, or detected only occasionally. >>> >>> On the same system, if I call into the voip number, calls from any of >>> the three external options (landline, cell, sip) all seem to work fine >>> with dtmf. >>> >>> So the problem seems to be something I've misconfigured with the tdm >>> card. Can anyone point me in the right direction? >> Hm. Completely the same problem. >> >> Can someone help? >> >> Here is the loghttp://pastebin.freeswitch.org/20447 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Tim Bock > Synacktics, LLC > www.synacktics.com > (505)795-1511 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/b049af2d/attachment.html From mike at jerris.com Fri Jan 18 01:23:38 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Jan 2013 17:23:38 -0500 Subject: [Freeswitch-users] 488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH In-Reply-To: <20130117112425.5ba534597f72c69dc122e214@yandex.ru> References: <20130117112425.5ba534597f72c69dc122e214@yandex.ru> Message-ID: FreeSWITCH does not currently support many new un-approved standards they are adding for WebRTC. If you are interested in contributing to the bounty for webrtc support you can email details to consulting at freeswitch.org. On Jan 17, 2013, at 2:24 AM, royj wrote: > > http://jira.freeswitch.org/browse/FS-4606 > > On Thu, 17 Jan 2013 02:05:34 +0530 > Paan Singh wrote: > >> New SIP capture og with FreeSWITCH log syntax highlighting : >> http://pastebin.freeswitch.com/20457 >> >> >> On Thu, Jan 17, 2013 at 12:50 AM, Paan Singh wrote: >> >>> Hi, >>> >>> I'm using JsSIP from a webpage to make a SIP call to FS, using OverSIP as >>> a Websocket->SIP proxy. >>> >>> I'm able to Register successfully, but when I make a call from JsSIP UA to >>> FreeSWITCH, I get a 180 Ringing, but after that I get a 488 Not Acceptable >>> here.. Please help me figure out what I'm doing wrong here. >>> >>> >>> My inbound_codec_prefs is : >>> >>> >>> >>> Here is FreeSWITCH's log for the incoming call : >>> >>> ============================================= >>> port.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN >>> tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) >>> tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg 0x2913be0 >>> from (udp/116.214.15.45:5080) has 3113 bytes, veclen = 1 >>> tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x2913be0 (3113 >>> bytes) from udp/121.245.35.150:5080/sip next=(nil) >>> nta.c:2803 agent_recv_request() nta: received INVITE >>> sip:30201009 at 116.214.15.45 SIP/2.0 (CSeq 2141) >>> nta.c:3161 agent_aliases() nta: canonizing sip:30201009 at 116.214.15.45with contact >>> nta.c:3002 agent_recv_request() nta: INVITE (2141) going to a default leg >>> nta.c:1348 set_timeout() nta: timer set to 200 ms >>> nua_server.c:102 nua_stack_process_request() nua: >>> nua_stack_process_request: entering >>> nua_stack.c:899 nh_create() nua: nh_create: entering >>> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering >>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering >>> soa.c:282 soa_clone() soa_clone(static::0x26122e0, 0x2606e20, >>> 0x7f91840c4c50) called >>> soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) >>> called >>> nta.c:4313 nta_leg_tcreate() nta_leg_tcreate(0x7f91840356d0) >>> soa.c:1302 soa_init_offer_answer() >>> soa_init_offer_answer(static::0x7f91840b25e0) called >>> soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7f91840b25e0, >>> (nil), 0x285676c, 2333) called >>> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f91840c4c50): adding >>> session usage >>> tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ >>> 121.245.35.150:5065 >>> tport.c:4026 tport_resolve() tport_resolve addrinfo = 121.245.35.150:5065 >>> tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not found >>> by name UDP/121.245.35.150:5065 >>> tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 564 bytes of 564 to udp/ >>> 121.245.35.150:5065 >>> tport.c:3472 tport_send_msg() tport_vsend returned 564 >>> nta.c:6678 incoming_reply() nta: sent 100 Trying for INVITE (2141) >>> nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_invite 100 >>> Trying >>> nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call >>> state changed: init -> received, received offer >>> soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7f91840b25e0, >>> [0x7f91aa98a5e8], [0x7f91aa98a5e0], [(nil)]) called >>> nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state 100 >>> Trying >>> nua_stack.c:359 nua_application_event() nua: nua_application_event: >>> entering >>> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering >>> 2013-01-17 00:25:27.733506 [NOTICE] switch_channel.c:968 New Channel >>> sofia/external/1000 at 116.214.15.45:5080[4aec7fd8-600e-11e2-8706-1365ba01ad52] >>> 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send signal >>> sofia/external/1000 at 116.214.15.45:5080 [BREAK] >>> nua_stack.c:359 nua_application_event() nua: nua_application_event: >>> entering >>> 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send signal >>> sofia/external/1000 at 116.214.15.45:5080 [BREAK] >>> 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_NEW >>> 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:433 >>> (sofia/external/1000 at 116.214.15.45:5080) State NEW >>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >>> 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5576 Channel sofia/external/ >>> 1000 at 116.214.15.45:5080 entering state [received][100] >>> 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5587 Remote SDP: >>> v=0 >>> o=- 2957375511 2 IN IP4 127.0.0.1 >>> s=- >>> t=0 0 >>> a=group:BUNDLE audio video >>> m=audio 40111 RTP/SAVPF 103 104 0 8 106 105 13 126 >>> c=IN IP4 121.245.35.150 >>> a=rtcp:40111 IN IP4 121.245.35.150 >>> a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host >>> generation 0 >>> a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host >>> generation 0 >>> a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx >>> generation 0 >>> a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx >>> generation 0 >>> a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host >>> generation 0 >>> a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host >>> generation 0 >>> a=ice-ufrag:qiKF0ErbxaXldKI0 >>> a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw >>> a=ice-options:google-ice >>> a=mid:audio >>> a=rtcp-mux >>> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >>> inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 >>> a=rtpmap:103 ISAC/16000 >>> a=rtpmap:104 ISAC/32000 >>> a=rtpmap:0 PCMU/8000 >>> a=rtpmap:8 PCMA/8000 >>> a=rtpmap:106 CN/32000 >>> a=rtpmap:105 CN/16000 >>> a=rtpmap:13 CN/8000 >>> a=rtpmap:126 telephone-event/8000 >>> a=ssrc:4278252985 cname:fMHr4uVHPcpHFfGu >>> a=ssrc:4278252985 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx >>> a=ssrc:4278252985 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx00 >>> m=video 40111 RTP/SAVPF 100 101 102 >>> c=IN IP4 121.245.35.150 >>> a=rtcp:40111 IN IP4 121.245.35.150 >>> a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host >>> generation 0 >>> a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host >>> generation 0 >>> a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx >>> generation 0 >>> a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx >>> generation 0 >>> a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host >>> generation 0 >>> a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host >>> generation 0 >>> a=ice-ufrag:qiKF0ErbxaXldKI0 >>> a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw >>> a=ice-options:google-ice >>> a=mid:video >>> a=rtcp-mux >>> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >>> inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 >>> a=rtpmap:100 VP8/90000 >>> a=rtpmap:101 red/90000 >>> a=rtpmap:102 ulpfec/90000 >>> a=ssrc:169275022 cname:fMHr4uVHPcpHFfGu >>> a=ssrc:169275022 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx >>> a=ssrc:169275022 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx10 >>> >>> 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 >>> AES_CM_128_HMAC_SHA1_80 inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2] >>> 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:3155 Set Local Key [1 >>> AES_CM_128_HMAC_SHA1_80 inline:i8vqmBeU4+/sjJBndCbEQ4G87gmpYeVnqxh8Xuzu] >>> 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:5277 No 2833 in SDP. >>> Disable 2833 dtmf and switch to INFO >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:2994 >>> (sofia/external/1000 at 116.214.15:45:5080) Callstate Change DOWN -> HANGUP >>> 2013-01-17 00:25:27.753511 [NOTICE] sofia.c:5878 Hangup sofia/external/ >>> 1000 at 116.214.15.45:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:3017 Send signal >>> sofia/external/1000 at 116.214.15.45:5080 [KILL] >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal >>> sofia/external/1000 at 116.214.15.45:5080 [BREAK] >>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_HANGUP >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 >>> (sofia/external/1000 at 116.214.15.45:5080) State HANGUP >>> 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:503 Channel sofia/external/ >>> 1000 at 116.214.15.45:5080 hanging up, cause: INCOMPATIBLE_DESTINATION >>> 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:633 Responding to INVITE >>> with: 488 >>> nua.c:879 nua_respond() nua: nua_respond: entering >>> nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_respond >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:48 >>> sofia/external/1000 at 116.214.15.45:5080 Standard HANGUP, cause: >>> INCOMPATIBLE_DESTINATION >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 >>> (sofia/external/1000 at 116.214.15.45:5080) State HANGUP going to sleep >>> nua_stack.c:573 nua_stack_signal() nua(0x7f91840c4c50): recv signal >>> r_respond 488 Not Acceptable Here >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:446 >>> (sofia/external/1000 at 116.214.15.45:5080) State Change CS_HANGUP -> >>> CS_REPORTING >>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal >>> sofia/external/1000 at 116.214.15.45:5080 [BREAK] >>> soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) >>> called >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_REPORTING >>> nua_session.c:2316 nua_invite_server_respond() nua: >>> nua_invite_server_respond: entering >>> soa.c:1214 soa_clear_remote_sdp() >>> soa_clear_remote_sdp(static::0x7f91840b25e0) called >>> tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ >>> 121.245.35.150:5065 >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 >>> (sofia/external/1000 at 116.214.15.45:5080) State REPORTING >>> tport.c:4026 tport_resolve() tport_resolve addrinfo = 121.245.35.150:5065 >>> tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not found >>> by name UDP/121.245.35.150:5065 >>> tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 784 bytes of 784 to udp/ >>> 121.245.35.150:5065 >>> tport.c:3472 tport_send_msg() tport_vsend returned 784 >>> nta.c:6678 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE >>> (2141) >>> nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f91840c4c50): >>> removing session usage >>> nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call >>> state changed: received -> terminated >>> nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state 488 >>> Not Acceptable Here >>> nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_terminated >>> 488 Not Acceptable Here >>> soa.c:358 soa_destroy() soa_destroy(static::0x7f91840b25e0) called >>> nta.c:4366 nta_leg_destroy() nta_leg_destroy(0x7f91840356d0) >>> nua_stack.c:359 nua_application_event() nua: nua_application_event: >>> entering >>> nua_stack.c:359 nua_application_event() nua: nua_application_event: >>> entering >>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >>> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering >>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering >>> nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering >>> nua_stack.c:569 nua_stack_signal() nua(0x7f91840c4c50): recv signal >>> r_destroy >>> nta.c:4366 nta_leg_destroy() nta_leg_destroy((nil)) >>> nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_destroy >>> 2013-01-17 00:25:27.753511 [DEBUG] mod_cdr_sqlite.c:102 Writing SQL to DB: >>> INSERT INTO cdr VALUES ("1000","1000","30201009","public","2013-01-17 >>> 00:25:27","","2013-01-17 >>> 00:25:27",0,0,"INCOMPATIBLE_DESTINATION","4aec7fd8-600e-11e2-8706-1365ba01ad52","","") >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:92 >>> sofia/external/1000 at 116.214.15.45:5080 Standard REPORTING, cause: >>> INCOMPATIBLE_DESTINATION >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 >>> (sofia/external/1000 at 116.214.15.45:5080) State REPORTING going to sleep >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/external/1000 at 116.214.15.45:5080) State Change CS_REPORTING -> >>> CS_DESTROY >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal >>> sofia/external/1000 at 116.214.15.45:5080 [BREAK] >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1499 Session 1794 >>> (sofia/external/1000 at 116.214.15.45:5080) Locked, Waiting on external >>> entities >>> 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1517 Session >>> 1794 (sofia/external/1000 at 116.214.15.45:5080) Ended >>> 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1521 Close >>> Channel sofia/external/1000 at 116.214.15.45:5080 [CS_DESTROY] >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:556 >>> (sofia/external/1000 at 116.214.15.45:5080) Callstate Change HANGUP -> DOWN >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:559 >>> (sofia/external/1000 at 116.214.15.45:5080) Running State Change CS_DESTROY >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 >>> (sofia/external/1000 at 116.214.15.45:5080) State DESTROY >>> 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:396 sofia/external/ >>> 1000 at 116.214.15.45:5080 SOFIA DESTROY >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:99 >>> sofia/external/1000 at 116.214.15.45:5080 Standard DESTROY >>> 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 >>> (sofia/external/1000 at 116.214.15.45:5080) State DESTROY going to sleep >>> nta.c:1294 agent_timer() nta: timer set next to 322 ms >>> tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN >>> tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) >>> tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg >>> 0x7f9198220070 from (udp/116.214.15.45:5080) has 339 bytes, veclen = 1 >>> tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x7f9198220070 >>> (339 bytes) from udp/121.245.35.150:5080/sip next=(nil) >>> nta.c:2803 agent_recv_request() nta: received ACK >>> sip:30201009 at 116.214.15.45 SIP/2.0 (CSeq 2141) >>> nta.c:2938 agent_recv_request() nta: ACK (2141) is going to INVITE (2141) >>> nta.c:1294 agent_timer() nta: timer set next to 4823 ms >>> nta.c:7010 _nta_incoming_timer() nta: timer I fired, terminate 488 response >>> nta.c:5718 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), >>> 0x7f91aa98ac20) >>> nta.c:7061 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, >>> 1/1 term, 1/1 free >>> nta.c:1279 agent_timer() nta: timer not set >>> ================================================================ >>> >>> Here is the Log from JsSIP in the browser that receives messages from >>> FreeSWITCH : >>> =========================================================== >>> JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 180 Ringing >>> Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 Record-Route: >>> Record-Route: >>> From: < >>> sip:1000 at 116.214.15.45:5080>;tag=rj2nc38qem To: < >>> sip:30201009 at 116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: >>> joshempnx2kuh7smrt3x CSeq: 1974 INVITE Contact: >>> User-Agent: >>> FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Accept: >>> application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, >>> UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, >>> replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 >>> P-Asserted-Identity: "30201009" >>> jssip-0.2.1.js:684 >>> JsSIP | DIALOG | New UAC dialog created: 1 jssip-0.2.1.js:2171 >>> JsSIP | EVENT EMITTER | Emitting event: progress jssip-0.2.1.js:170 >>> JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 488 Not >>> Acceptable Here Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 >>> Max-Forwards: 10 From: ;tag=rj2nc38qem To: < >>> sip:30201009 at 116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: >>> joshempnx2kuh7smrt3x CSeq: 1974 INVITE User-Agent: >>> FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Allow: INVITE, >>> ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY >>> Supported: timer, precondition, path, replaces Allow-Events: talk, hold, >>> conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>> Content-Length: 0 P-Asserted-Identity: "30201009" < >>> sip:30201009 at 116.214.15.45> >>> ============================================================================= >>> From mike at jerris.com Fri Jan 18 01:27:26 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Jan 2013 17:27:26 -0500 Subject: [Freeswitch-users] Mod_Conference Improvement (Denoise) In-Reply-To: <1358404199.68203.YahooMailNeo@web162901.mail.bf1.yahoo.com> References: <5274F7F1-8BC9-4721-B98F-A163495CD730@jerris.com> <1358404199.68203.YahooMailNeo@web162901.mail.bf1.yahoo.com> Message-ID: <641E563A-0AE5-4DFB-A1E8-714E0B40A011@jerris.com> If you are going to do any noise filtering of a call it should be done prior to messing with the volume such as agc. On Jan 17, 2013, at 1:29 AM, Valer Nur wrote: > I think the best way to solve this is by doing AGC first. This should make sure audio level is similar for all participants. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/a1901c28/attachment.html From dvl36.ripe.nick at gmail.com Fri Jan 18 02:41:58 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Fri, 18 Jan 2013 01:41:58 +0200 Subject: [Freeswitch-users] Linux PREEMPT RT kernel and freeswitch -rp Message-ID: Hello, freeswitchers. Does anybody here use freeswitch -rp with PREEMPT RT Linux kernel? WBR. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/0d0b1912/attachment.html From admin at smallunix.net Fri Jan 18 02:59:19 2013 From: admin at smallunix.net (Andrea Mazzeo) Date: Fri, 18 Jan 2013 00:59:19 +0100 Subject: [Freeswitch-users] SONUS - dtg inside INVITE In-Reply-To: References: Message-ID: <7443520999689776200@unknownmsgid> Hi Kristian, I need dtg in the INVITE header. Best regards, Andrea Mazzeo Il giorno 17/gen/2013, alle ore 22:01, Kristian Kielhofner ha scritto: > This is almost certainly not the syntax you want to use. > > Do that expect "dtg" to be a user or domain parameter? > > On Thu, Jan 17, 2013 at 1:46 PM, Andrea Mazzeo wrote: >> Hi all, >> >> I need to configure my FS to send call to a Sonus GSX via SIP. >> Is there a way to send SIP parameter "dtg" (destination trunk group) >> inside the INVITE? >> >> Example: >> >> Request-Line: INVITE sip: 39340xxxxxx at 38.xxx.xxx.xxx SIP/2.0 dtg=ATTMDT7807IPO >> >> >> Thanks, >> Andrea >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Fri Jan 18 03:31:24 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 17 Jan 2013 19:31:24 -0500 Subject: [Freeswitch-users] SONUS - dtg inside INVITE In-Reply-To: <7443520999689776200@unknownmsgid> References: <7443520999689776200@unknownmsgid> Message-ID: <5D31EF90-20B7-48BA-B7BC-65F30B5383DD@jerris.com> What you posted is NOT a valid invite header. WHERE in the invite header do you need it? On Jan 17, 2013, at 6:59 PM, Andrea Mazzeo wrote: > Hi Kristian, > > I need dtg in the INVITE header. > > Best regards, > Andrea Mazzeo > > Il giorno 17/gen/2013, alle ore 22:01, Kristian Kielhofner > ha scritto: > >> This is almost certainly not the syntax you want to use. >> >> Do that expect "dtg" to be a user or domain parameter? >> >> On Thu, Jan 17, 2013 at 1:46 PM, Andrea Mazzeo wrote: >>> Hi all, >>> >>> I need to configure my FS to send call to a Sonus GSX via SIP. >>> Is there a way to send SIP parameter "dtg" (destination trunk group) >>> inside the INVITE? >>> >>> Example: >>> >>> Request-Line: INVITE sip: 39340xxxxxx at 38.xxx.xxx.xxx SIP/2.0 dtg=ATTMDT7807IPO >>> From steveayre at gmail.com Fri Jan 18 04:57:36 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Jan 2013 01:57:36 +0000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> Message-ID: > What're you hoping to do here? This sets the payload type value. 0 is a valid pt. This won't disable it, it'll instead use the value 0. That happens to be the one for G711, so if that codec is enabled you're just going to get a conflict. More likely you want: > This is also probably not a good idea. These are attributes of specific codecs and those modules will set them if required. FS is negotiating and terminating media, you shouldn't be needing to play with the SDP at a low level for what you're trying to do. Steve on iPhone On 17 Jan 2013, at 02:58, "support at ecn.net.au" wrote: > Hi > > This sounds quite posible. > > I've tcpdumped sip headers on both the freeswitch and the backend PBX. > > The only significant difference I can see between our old SBC and FS is that the old SBC > asserted a media attribute fmtp:18 annexb=no where as FS doesn't forward this. > > To test it out we have set dtmf_mode to none on both contexts (the context facing the telco > and the context facing the pbx), and set > > > > (this may not be needed). > > then in the dialplan we have > > > > set prior to the bridge. > > -- > > When we sip trace however we are not appending the attribute to the SDP on the INVITE. > > Logs and sip headers below: > > > EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060 export(sip_append_audio_sdp=a=fmtp:18 annexb=no) > 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] > EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060 bridge(sofia/external/0737111111 at A.B.C.X:5060) > 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1089 sofia/aapt/0731111111 at X.Y.Z.A:5060 EXPORTING[export_vars] [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to eve > nt > 2013-01-17 12:34:19.702102 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables > 2013-01-17 12:34:19.702102 [NOTICE] switch_channel.c:968 New Channel sofia/external/0737111111 at A.B.C.X:5060 [a30c650a-87d3-4027-9010-213547d698aa] > 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:4977 (sofia/external/0737111111 at A.B.C.X:5060) State Change CS_NEW -> CS_INIT > 2013-01-17 12:34:19.702102 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/0737111111 at A.B.C.X:5060 [BREAK] > 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0737111111 at A.B.C.X:5060) Running State Change CS_INIT > 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:454 (sofia/external/0737111111 at A.B.C.X:5060) State INIT > 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:86 sofia/external/0737111111 at A.B.C.X:5060 SOFIA INIT > 2013-01-17 12:34:19.702102 [DEBUG] sofia_glue.c:2647 Local SDP: > v=0 > o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D > s=FreeSWITCH > c=IN IP4 A.B.C.D > t=0 0 > m=audio 24552 RTP/AVP 8 0 3 13 > a=fmtp:18 annexb=no > a=ptime:20 > a=sendrecv > > > But then the actual Invite doesn't contain the attribute in the SDP > > send 1036 bytes to udp/[A.B.C.X]:5060 at 02:34:19.718385: > ------------------------------------------------------------------------ > INVITE sip:0737111111 at A.B.C.X:5060 SIP/2.0 > Via: SIP/2.0/UDP A.B.C.D:5080;rport;branch=z9hG4bKpUvH6K1ppUSNp > Max-Forwards: 69 > From: "0731111111" ;tag=mH93ZHryv4H2g > To: > Call-ID: 3ca1df3f-daf1-1230-f389-002219a7c712 > CSeq: 38858965 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.2.5.3+git~20121229T001759Z~e04eab7902 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 155 > X-Nortel-Profile: DEFAULT > X-FS-Support: update_display,send_info > Remote-Party-ID: "0731111111" ;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D > s=FreeSWITCH > c=IN IP4 A.B.C.D > t=0 0 > m=audio 24552 RTP/AVP 8 0 3 13 > a=ptime:20 > ------------------------------------------------------------------------ > > > > Are we missing something do you think? > > > > Kind Regards, > > From: Steven Ayre [steveayre at gmail.com] > Sent: Thursday, 17 January 2013 4:14 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SBC In-band DTMF > >> However under freeswitch if we don't start_dtmf before the bridge the backend PBX boxes don't recognise >> the DTMF inband (even though the tones are audible ie you can hear them on a call recording on the >> PBX). >> >> Have we missed something here? We would have thought with inband DTMF on non compressed codec (no >> transcoding) that the tones would just work with the media stream? > > start_dtmf will detect inband DTMF, and then send out of band on the outgoing leg of the bridge. This is expected. You'll will need to call start_dtmf from dialplan, which can easily be done from dialplan by checking for some condition identifying calls requiring it. If they're authenticating to FS then it'd be trivial to set a variable in their user directory entry that you can test in the dialplan. > > Now this is just a guess... During the codec negotiation FS will tell your PBX that it supports telephone-event (RFC2833). It may be that because the PBX sees that in the SDP that it doesn't look for inband DTMF, while when it received the SDP direct from the customer it didn't contain telephone-event and so did check for inband DTMF. Indeed it'd probably be a good idea for them to do so, since if you receive the same DTMF digit both inband and through RFC2833 then you'd be duplicating digits. So my guess is that the PBX doesn't bother check for inband because FS tells it it's sending out-of-band, which'd differ from before. > > -Steve > > > > -Steve > > > > > On 16 January 2013 04:02, support at ecn.net.au wrote: > > Hi All > > > > > > > > We're quite new to Freeswitch and are in the process of migrating from > > OpenSer (as an SBC) to Freeswitch. > > > > > > > > Mostly all is working well, except an oddity on DTMF. > > > > > > > > Our scenario: > > > > > > > > Telco/SIP Provider A is passing us calls using DTMF inband. > > > > > > > > We have a freeswitch configured as a SBC using 2 sip profiles (telco and > > internal) to topology hide and manage > > > > distribution of calls to the PBX servers located behind the SBC. > > > > > > > > The freeswitch will be handling up to a few hundred calls so we're trying to > > keep it lightweight. > > > > > > > > Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling > > customer needs. > > > > > > > > An example inbound call profile looks like this: > > > > > > > > > > > > > > > > > > > > > > > > > data="sofia/external/123456 at INTERNAL.PBX.IP:5060"/> > > > > > > > > > > > > > > > > Initially when calling into the platform IVR type applications runinng on > > our PBX boxes would not > > > > work (you could hear the DTMF but the platform did not recognise the tones). > > > > > > > > We have had to add the appliation start_dtmf in order for Freeswitch to pass > > the DTMF to the Asterisk > > > > PBX behind the SBC. Interestingly on our OpenSer platform we just proxied > > the media (rtpproxy) with > > > > inband DTMF from the Telco and our PBX boxes recognised the inband DTMF > > tones on the PBX platforms and > > > > IVR type applications just worked. > > > > > > > > However under freeswitch if we don't start_dtmf before the bridge the > > backend PBX boxes don't recognise > > > > the DTMF inband (even though the tones are audible ie you can hear them on a > > call recording on the > > > > PBX). > > > > > > > > Have we missed something here? We would have thought with inband DTMF on > > non compressed codec (no > > > > transcoding) that the tones would just work with the media stream? > > > > > > > > We have confirmed both legs are PCMA and when using start_dtmf the first > > second of the call is clipped. > > > > > > > > > > > > > > > > Kind Regards, > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/c05ed695/attachment-0001.html From steveayre at gmail.com Fri Jan 18 05:07:31 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Jan 2013 02:07:31 +0000 Subject: [Freeswitch-users] limit memory usage of freeswitch In-Reply-To: <17844111.8921.1358419535248.JavaMail.javamailuser@localhost> References: <17844111.8921.1358419535248.JavaMail.javamailuser@localhost> Message-ID: That version is absolutely ancient. It is possible you got a revision that contains a memory leak that has since been solved. Absolutely try to reproduce it on v1.2.stable, you'll probably find it just goes away. If it *is* still happening the 2011-10 version will be too old to support (code will have changed too much) but with git head of v1.2.stable or master the devs will be able to investigate. Steve on iPhone On 17 Jan 2013, at 10:45, Raimund Sacherer wrote: > Hello Anthony, > > I also see Freeswitch eating away on Memory. Right now we are in the process of installing a new FS Cluster and I will upgrade all our PBX's with the new stable branch and I will test with that again. > > Our version of Freeswitch in production is: FreeSWITCH Version 1.0.head (git-4ceca18 2011-10-21 20-38-41 -0500) > This version slowly grows and grows, Over month the biggest grow I saw was about 4-5 Gigabyte of RES memory. > > Before that we used a Freeswitch version from 2011-03 and that version grow to 273 Megabyte of memory usage and did not grow further. > > In the future I will report my findings with the new stable branch, hopefully whatever caused the memory consumption is allready fixed. > > Best regards, > Ray > > From: "Anthony Minessale" > To: "FreeSWITCH Users Help" > Sent: Jueves, 17 de Enero 2013 2:12:58 > Subject: Re: [Freeswitch-users] limit memory usage of freeswitch > > It does not eat memory.... if you attack it with sipp you may cause the internal pools to swell up but normal usage only consumes memory based on usage. > > > On Wed, Jan 16, 2013 at 4:09 AM, jay binks wrote: >> OpenVZ / Proxmox will allow you use set quota's and containerize your freeswitch... >> thats ONE way to do what your after. >> >> >> On 16 January 2013 19:48, rentmycoder rentmycoder wrote: >>> Hi All, >>> >>> According to my experineces and some info found on FS lists it is the normal behaviour that freeswitch eats up all of the available memory... >>> It's not a memory leak, but the standard behaviour of FS by design... >>> Correct me if I'm wrong... >>> >>> Is there any way to limit FS memory usage??? >>> >>> I need another services to run on the FS box... >>> >>> Thanks, >>> John >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Sincerely >> >> Jay >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/d1d50826/attachment.html From steveayre at gmail.com Fri Jan 18 05:23:11 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Jan 2013 02:23:11 +0000 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> Message-ID: The latest version of opal in Debian (even in Sid) isn't new enough for mod_opal. It needed various features added that are only only available in the svn version. The wiki page states you have to install ptlib and opal from svn - this is why. http://wiki.freeswitch.org/wiki/Mod_opal Of course that can then cause conflicts unless you package it yourself or install a dummy package, or problems with other libraries/apps that are built against the libopal that's already packaged. Ptlib and h323plus/opal are also notorious for having to get matching versions. If you don't pair exactly the right version of ptlib with the specific version of h323plus/opal you can get instability problems. So the official Debian ptlib package probably wouldn't be good with opal from svn... As for mod_h323, it does compile and work, at least on Wheezy and possibly Squeeze too. I've never had any luck getting mod_opal working but have got mod_h323 working. I don't actively use it though - all my traffic is currently SIP. Another option is to run Yate as a SIP->H323 proxy. It only handles the signalling, media still goes directly caller->FS. It worked quite well, although it would occasionally crash - but at least that only dropped the H323 calls and didn't take FS down with it, and Yate automatically restarted. That was on Lenny so a long time ago... Steve on iPhone On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote: > Hi, > > I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and opal > 3.10.7 on a debian > Ptlib and opal get build with no error, but when trying mod_opal I get > > Compiling > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... > quiet_libtool: compile: g++ -g -ggdb > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src > -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DP_64BIT > -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions > -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL > -DHAVE_CONFIG_H -c > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC > -DPIC -o .libs/mod_opal.o > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member > function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, > switch_frame_t**, switch_io_flag_t)': > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: > error: 'class OpalCall' has no member named 'IsSwitchingT38' > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member > function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, > const switch_frame_t*, switch_io_flag_t)': > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: > error: 'class OpalCall' has no member named 'IsSwitchingT38' > make[3]: *** [mod_opal.lo] Error 1 > make[2]: *** [install] Error 1 > make[1]: *** [mod_opal-install] Error 1 > make: *** [mod_opal-install] Error 2 > > > Has anyone succeded in building mod_opal ? > > Should I go with mod_h323 instead ? > > Thanks for your help > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/109349d3/attachment-0001.html From krice at freeswitch.org Fri Jan 18 05:31:11 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 17 Jan 2013 20:31:11 -0600 Subject: [Freeswitch-users] limit memory usage of freeswitch In-Reply-To: Message-ID: Stable nothing... Reproduce on head.... On 1/17/13 8:07 PM, "Steven Ayre" wrote: > That version is absolutely ancient. It is possible you got a revision that > contains a memory leak that has since been solved. Absolutely try to reproduce > it on v1.2.stable, you'll probably find it just goes away. > > If it *is* still happening the 2011-10 version will be too old to support > (code will have changed too much) but with git head of v1.2.stable or master > the devs will be able to investigate. > > Steve on iPhone > > > On 17 Jan 2013, at 10:45, Raimund Sacherer > wrote: > >> Hello Anthony, >> >> I also see Freeswitch eating away on Memory. Right now we are in the process >> of installing a new FS Cluster and I will upgrade all our PBX's with the new >> stable branch and I will test with that again. >> >> Our version of Freeswitch in production is: FreeSWITCH Version 1.0.head >> (git-4ceca18 2011-10-21 20-38-41 -0500) >> This version slowly grows and grows, Over month the biggest grow I saw was >> about 4-5 Gigabyte of RES memory. >> >> Before that we used a Freeswitch version from 2011-03 and that version grow >> to 273 Megabyte of memory usage and did not grow further. >> >> In the future I will report my findings with the new stable branch, hopefully >> whatever caused the memory consumption is allready fixed. >> >> Best regards, >> Ray >> >> >> From: "Anthony Minessale" >> To: "FreeSWITCH Users Help" >> Sent: Jueves, 17 de Enero 2013 2:12:58 >> Subject: Re: [Freeswitch-users] limit memory usage of freeswitch >> >> It does not eat memory.... if you attack it with sipp you may cause the >> internal pools to swell up but normal usage only consumes memory based on >> usage. >> >> >> On Wed, Jan 16, 2013 at 4:09 AM, jay binks wrote: >>> OpenVZ / Proxmox will allow you use set quota's and containerize your >>> freeswitch... >>> thats ONE way to do what your after. >>> >>> >>> On 16 January 2013 19:48, rentmycoder rentmycoder >>> wrote: >>>> Hi All, >>>> >>>> According to my experineces and some info found on FS lists it is the >>>> normal behaviour that freeswitch eats up all of the available memory... >>>> It's not a memory leak, but the standard behaviour of FS by design... >>>> Correct me if I'm wrong... >>>> >>>> Is there any way to limit FS memory usage??? >>>> >>>> I need another services to run on the FS box... >>>> >>>> Thanks, >>>> John >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/5471a718/attachment.html From support at ecn.net.au Fri Jan 18 08:19:31 2013 From: support at ecn.net.au (support at ecn.net.au) Date: Fri, 18 Jan 2013 15:19:31 +1000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting>, Message-ID: <8A80C99638A5E24E83696EAE33A9B6B968B5302818@twbexmbx001.ecn.hosting> Thanks for the replies so far, your input is greatly appreciated Our configuration is TELCO --SIP--> FREESWITCH/SBC --SIP--> PBX Our Telco uses Inband DTMF for tones. Our legacy SBC platform (openser) simply proxied the content through and the PBX's recognised the inband dtmf tones and acted accordingly. Since moving onto FreeSwitch an SBC DTMF on calls made from the telco through to a PBX are not being recognised by the PBX as inband (ie no DTMF). A few notes: 1) We can add start_dtmf as an application prior to bridge and set dtmf_mode none in the Telco facing context and dtmf_mode rfc2833 in the pbx facing context, and DTMF is recognised on the PBX (as rfc2833), However we had a resultant issue that audio file playback from IVR's on the Asterisk PBX is clipped (first 2 seconds or so of each sound file being played and this does not occur on voice mail box recordings - tested with the same audio file) - interestingly on Freeswitch 1.0.6 this clipping does not occur, on 1.2.x or the unstable 1.3 every version exhibits this odd behaviour. (Sound from remote IVR on PBX clipped). An interesting notes using the record session application, the clipping is not apparent, only to the calling party. (Could this possibly be legitimate bug that should be submitted?) So we considered that we should be able to just pass the dtmf inband without looking at the media, however so far we can not seem to make the PBX's recognise DTMF inband on inbound sip calls. The earlier post http://lists.freeswitch.org/pipermail/freeswitch-users/2013-January/091332.html suggested to look at the way freeswitch was setting up the invite to the PBX (perhaps not telling the PBX that we are using inband). So we tcpdumped the SIP INVITES from the Telco to the SBC and then from the SBC to the PBX, and also did the same on our legacy openser SBC as well. the only defined difference that we could ascertain was that when using Freeswitch as the SBC the fmtp:18 media attribute (present between on the telco's invite on the a leg, and present on the openser->pbx invite on the legacy is missing on the Freeswitch Invite to the PBX. Reading suggested to add the directive to add this media attribute to the b-leg SDP. However (and we may be missing the point here) when we do this it has no impact on the SDP INVITE sent from the Freeswitch to the PBX (using tcddump to capture the network traffic). -- It seems we have 2 issues; the First is why does start_dtmf in freeswitch 1.2.x and 1.3.x clips the sounds (when in 1.0.x it does not) the Second is why doesn't adding the sip_append_audio_sdp actually add the attribute or how can we make Freeswitch do this (it appears in reading fmtp:18 and dtmf inband have some relationship, we've made an assumption that this may be why the PBX's are not looking for inband dtmf on these calls. any help is really appreciated as we're currently stuck until we figure this out! Kind Regards, ________________________________ From: Steven Ayre [steveayre at gmail.com] Sent: Friday, 18 January 2013 11:57 AM To: FreeSWITCH Users Help Cc: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SBC In-band DTMF What're you hoping to do here? This sets the payload type value. 0 is a valid pt. This won't disable it, it'll instead use the value 0. That happens to be the one for G711, so if that codec is enabled you're just going to get a conflict. More likely you want: This is also probably not a good idea. These are attributes of specific codecs and those modules will set them if required. FS is negotiating and terminating media, you shouldn't be needing to play with the SDP at a low level for what you're trying to do. Steve on iPhone On 17 Jan 2013, at 02:58, "support at ecn.net.au" > wrote: Hi This sounds quite posible. I've tcpdumped sip headers on both the freeswitch and the backend PBX. The only significant difference I can see between our old SBC and FS is that the old SBC asserted a media attribute fmtp:18 annexb=no where as FS doesn't forward this. To test it out we have set dtmf_mode to none on both contexts (the context facing the telco and the context facing the pbx), and set (this may not be needed). then in the dialplan we have set prior to the bridge. -- When we sip trace however we are not appending the attribute to the SDP on the INVITE. Logs and sip headers below: EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060 export(sip_append_audio_sdp=a=fmtp:18 annexb=no) 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060 bridge(sofia/external/0737111111 at A.B.C.X:5060) 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1089 sofia/aapt/0731111111 at X.Y.Z.A:5060 EXPORTING[export_vars] [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to eve nt 2013-01-17 12:34:19.702102 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-01-17 12:34:19.702102 [NOTICE] switch_channel.c:968 New Channel sofia/external/0737111111 at A.B.C.X:5060 [a30c650a-87d3-4027-9010-213547d698aa] 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:4977 (sofia/external/0737111111 at A.B.C.X:5060) State Change CS_NEW -> CS_INIT 2013-01-17 12:34:19.702102 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/0737111111 at A.B.C.X:5060 [BREAK] 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0737111111 at A.B.C.X:5060) Running State Change CS_INIT 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:454 (sofia/external/0737111111 at A.B.C.X:5060) State INIT 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:86 sofia/external/0737111111 at A.B.C.X:5060 SOFIA INIT 2013-01-17 12:34:19.702102 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D s=FreeSWITCH c=IN IP4 A.B.C.D t=0 0 m=audio 24552 RTP/AVP 8 0 3 13 a=fmtp:18 annexb=no a=ptime:20 a=sendrecv But then the actual Invite doesn't contain the attribute in the SDP send 1036 bytes to udp/[A.B.C.X]:5060 at 02:34:19.718385: ------------------------------------------------------------------------ INVITE sip:0737111111 at A.B.C.X:5060 SIP/2.0 Via: SIP/2.0/UDP A.B.C.D:5080;rport;branch=z9hG4bKpUvH6K1ppUSNp Max-Forwards: 69 From: "0731111111" ;tag=mH93ZHryv4H2g To: Call-ID: 3ca1df3f-daf1-1230-f389-002219a7c712 CSeq: 38858965 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.5.3+git~20121229T001759Z~e04eab7902 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 155 X-Nortel-Profile: DEFAULT X-FS-Support: update_display,send_info Remote-Party-ID: "0731111111" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D s=FreeSWITCH c=IN IP4 A.B.C.D t=0 0 m=audio 24552 RTP/AVP 8 0 3 13 a=ptime:20 ------------------------------------------------------------------------ Are we missing something do you think? Kind Regards, ________________________________ From: Steven Ayre [steveayre at gmail.com] Sent: Thursday, 17 January 2013 4:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SBC In-band DTMF However under freeswitch if we don't start_dtmf before the bridge the backend PBX boxes don't recognise the DTMF inband (even though the tones are audible ie you can hear them on a call recording on the PBX). Have we missed something here? We would have thought with inband DTMF on non compressed codec (no transcoding) that the tones would just work with the media stream? start_dtmf will detect inband DTMF, and then send out of band on the outgoing leg of the bridge. This is expected. You'll will need to call start_dtmf from dialplan, which can easily be done from dialplan by checking for some condition identifying calls requiring it. If they're authenticating to FS then it'd be trivial to set a variable in their user directory entry that you can test in the dialplan. Now this is just a guess... During the codec negotiation FS will tell your PBX that it supports telephone-event (RFC2833). It may be that because the PBX sees that in the SDP that it doesn't look for inband DTMF, while when it received the SDP direct from the customer it didn't contain telephone-event and so did check for inband DTMF. Indeed it'd probably be a good idea for them to do so, since if you receive the same DTMF digit both inband and through RFC2833 then you'd be duplicating digits. So my guess is that the PBX doesn't bother check for inband because FS tells it it's sending out-of-band, which'd differ from before. -Steve -Steve On 16 January 2013 04:02, support at ecn.net.au > wrote: > Hi All > > > > We're quite new to Freeswitch and are in the process of migrating from > OpenSer (as an SBC) to Freeswitch. > > > > Mostly all is working well, except an oddity on DTMF. > > > > Our scenario: > > > > Telco/SIP Provider A is passing us calls using DTMF inband. > > > > We have a freeswitch configured as a SBC using 2 sip profiles (telco and > internal) to topology hide and manage > > distribution of calls to the PBX servers located behind the SBC. > > > > The freeswitch will be handling up to a few hundred calls so we're trying to > keep it lightweight. > > > > Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling > customer needs. > > > > An example inbound call profile looks like this: > > > > > > > > > > > > data="sofia/external/123456 at INTERNAL.PBX.IP:5060"/> > > > > > > > > Initially when calling into the platform IVR type applications runinng on > our PBX boxes would not > > work (you could hear the DTMF but the platform did not recognise the tones). > > > > We have had to add the appliation start_dtmf in order for Freeswitch to pass > the DTMF to the Asterisk > > PBX behind the SBC. Interestingly on our OpenSer platform we just proxied > the media (rtpproxy) with > > inband DTMF from the Telco and our PBX boxes recognised the inband DTMF > tones on the PBX platforms and > > IVR type applications just worked. > > > > However under freeswitch if we don't start_dtmf before the bridge the > backend PBX boxes don't recognise > > the DTMF inband (even though the tones are audible ie you can hear them on a > call recording on the > > PBX). > > > > Have we missed something here? We would have thought with inband DTMF on > non compressed codec (no > > transcoding) that the tones would just work with the media stream? > > > > We have confirmed both legs are PCMA and when using start_dtmf the first > second of the call is clipped. > > > > > > > > Kind Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/54c0cbcc/attachment-0001.html From craig.robert at hotmail.com Fri Jan 18 08:32:28 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Thu, 17 Jan 2013 22:32:28 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: , , , , , , , , , , , , , , , , , Message-ID: OK, so I've made a little bit of progress. The distant end is a Windows Fax server hooked up to an analog line. I created a fax extension in FS, assigned an inbound route to the extension, and made sure the extension had an email address. I believe the fax receive wants to complete, but there is an interesting error in the log about TIF files. Anyone ever seen this before? v=0 o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 s=FreeSWITCH c=IN IP4 72.201.112.131 t=0 0 m=image 27714 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:2000 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === Negotiation Result ======================================================= 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote station id: ATW-4792460547 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local station id: SpanDSP Fax Ident 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer Rate: 9600 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM status off 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote country: 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote vendor: 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote model: 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 ============================================================================== 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (41) TIFF/F file cannot be opened. 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote station id: ATW-4792460547 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 9600 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM status off 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote country: 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote vendor: 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote model: 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. EXECUTE sofia/external/+14792460547 at flowroute.com hangup() Robert From: craig.robert at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 16 Jan 2013 10:40:31 -0700 Subject: Re: [Freeswitch-users] T38 Configuration OK, I am not extremely concerned with getting an ATA working right now. If I could just get an inbound extension-to-email working it would be great. I'll start from scratch and then let you know what happens. Just to make sure, I am going to create an extension, assign an email address to it, make sure there is an inbound route to that extension, and then test it. Robert Date: Wed, 16 Jan 2013 12:11:34 -0500 From: gassaad at emassembly.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration The reason for outbound extension is:When you send a fax from ATA to the outside world, it will dial first to the outbound extension and then after successful receive it will dial out to the intended number. As for the error you have it may be related to your ATA. I am not sure please see why do you get the following message."Invalid Jitterbuffer spec [0] must be between 20 and 10000 " On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig wrote: OK, I've already tried creating one extension for inbound and set an email address. Of course, it failed and the errors are at the bottom of this email. What is the purpose of the outbound extension? Robert Date: Wed, 16 Jan 2013 09:57:51 -0500 From: gassaad at emassembly.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Sorry for the delay to get back to you. 1- App --> fax server2- Create two extensions (one for inbound) another for (outbound) From within either extension you can send faxes "virtual fax" 3- point your create your outbound and inbound dialplans as normal but make it to point to those extensions. Good luck, George On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig wrote: What did you do on the config side to get it to work? I've tried both an Ata and a fax to email extension and they both fail in both directions. Robert On Jan 12, 2013, at 2:04 PM, "George Assaad" wrote: I use Fusion successfully to send / receive fax.The way it works: ATA --->FaxServer.After successful receipt, Fax Server---> DID where you have dialled originally. Cheers, George On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig wrote: Did anyone see the issue or what I may have missed in the config? As mentioned earlier, I thought it would after a fresh install. Any help is appreciated. Robert From: craig.robert at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Thu, 10 Jan 2013 13:43:09 -0700 Subject: Re: [Freeswitch-users] T38 Configuration Thanks for the quick response. Below is the capture from the log viewer. In this test, I am using a remote fax machine (line from Cox with physical machine) faxing into the "email server" where I created an extension for fax-to-email. Calling number is 1479-246-0547. 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 sofia/external/+14792460547 at flowroute.com Standard DESTROY 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/+14792460547 at flowroute.com SOFIA DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> DOWN 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 (sofia/external/+14792460547 at flowroute.com) Ended 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external entities 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> CS_DESTROY 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING going to sleep 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: NORMAL_CLEARING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> CS_REPORTING 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup Command with no Session system(/usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 caller_id_name='' caller_id_number=): 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: NORMAL_CLEARING 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to sofia/external/+14792460547 at flowroute.com 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel sofia/external/+14792460547 at flowroute.com hanging up, cause: NORMAL_CLEARING 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_HANGUP 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 sofia/external/+14792460547 at flowroute.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal sofia/external/+14792460547 at flowroute.com [KILL] 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP EXECUTE sofia/external/+14792460547 at flowroute.com hangup() 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 14400 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station id: ATW-4792460547 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (48) Disconnected after permitted retries. 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer spec [0] must be between 20 and 10000 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxRateManagement:transferredTCF m=image 31632 udptl t38 t=0 0 c=IN IP4 66.42.121.5 s=- o=- 275426305 1357849876 IN IP4 66.42.121.5 v=0 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completing][200] 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+14792460547 at flowroute.com Update Callee ID to "+14792460547" <+14792460547> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [calling][0] 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxMaxDatagram:400 a=T38FaxMaxBuffer:2000 a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval a=T38MaxBitRate:9600 a=T38FaxVersion:0 m=image 27588 udptl t38 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec activation Success L16 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 sofia/external/+14792460547 at flowroute.com Push codec L16:70 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec activation Success L16 20000 EXECUTE sofia/external/+14792460547 at flowroute.com rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done playing file silence_stream://2000 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms EXECUTE sofia/external/+14792460547 at flowroute.com playback(silence_stream://2000) 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [last_fax]=[+14792460547-2013-01-10-13-39-26] EXECUTE sofia/external/+14792460547 at flowroute.com set(last_fax=+14792460547-2013-01-10-13-39-26) 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38_request=true) 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38]=[true] 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completed][200] EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38=true) 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel [sofia/external/+14792460547 at flowroute.com] has been answered 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] a=sendrecv a=ptime:20 a=silenceSupp:off - - - - a=fmtp:101 0-16 a=rtpmap:101 telephone-event/8000 a=rtpmap:0 PCMU/8000 m=audio 27588 RTP/AVP 0 101 t=0 0 c=IN IP4 72.201.112.131 s=FreeSWITCH o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 v=0 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP sofia/external/+14792460547 at flowroute.com: 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success [72.201.112.131]:[27588] EXECUTE sofia/external/+14792460547 at flowroute.com answer() 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=${last_fax} messages='result: ${fax_result_text} sender:${fax_remote_station_id} pages:${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] EXECUTE sofia/external/+14792460547 at flowroute.com set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} messages='result: \${fax_result_text} sender:\${fax_remote_station_id} pages:\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/+14792460547/3150) EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [call_direction]=[local] EXECUTE sofia/external/+14792460547 at flowroute.com set(call_direction=local) 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() Dialplan: sofia/external/+14792460547 at flowroute.com Action rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) Dialplan: sofia/external/+14792460547 at flowroute.com Action playback(silence_stream://2000) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38_request=true) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38=true) Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() Dialplan: sofia/external/+14792460547 at flowroute.com Action set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} pages:\\\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\\\${caller_id_name}' caller_id_number=\\\${caller_id_number}) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test Fax] destination_number(3150) =~ /^3150$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Test Fax] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain_user] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->send_to_voicemail] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->extension-intercom] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->intercept-ext] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_return] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_privacy] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [global] Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->global] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [redial] destination_number(3150) =~ /^(redial|\*870)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->redial] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->group-intercept] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action set(call_direction=local) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call-direction] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Flowroute.11d] continue=false 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->3150 in context default 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer sofia/external/+14792460547 at flowroute.com to XML[3150 at default] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML default) 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] EXECUTE sofia/external/+14792460547 at flowroute.com export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [outside_call]=[true] EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 XML default) Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] context(public) =~ /public/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->Test_Number] continue=false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->call_debug] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/+14792460547 at flowroute.com Action set(outside_call=true) Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [outside_call] Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->outside_call] continue=true Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->unloop] continue=false 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->15203039810 in context public 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+14792460547 at flowroute.com) Callstate Change DOWN -> RINGING 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+14792460547 at flowroute.com SOFIA INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf send/recv payload to 101 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 sofia/external/+14792460547 at flowroute.com Original read codec set to PCMU:0 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples 64000 bits 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] a=ptime:20 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=rtpmap:18 G729/8000 m=audio 31632 RTP/AVP 0 18 101 t=0 0 c=IN IP4 66.42.121.5 s=- o=- 275426305 1357849875 IN IP4 66.42.121.5 v=0 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [received][100] 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 (sofia/external/+14792460547 at flowroute.com) State NEW 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel sofia/external/+14792460547 at flowroute.com [d276f012-5b65-11e2-beae-8564ff827a25] Robert Date: Thu, 10 Jan 2013 09:47:18 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Can you pastebin a console debug log w/ sip trace of a call failing to negotiate t.38 correctly? That might help us figure out where to look. -MC On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: Just installed FusionPBX using the script built for a Centos 6 install. Everything works great in regards to calls, but my main goal of this project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have tried many things in media bypass, routing to a fax extension, using an ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me in the right direction as to where I should begin? I am guessing it isn't supposed to work right out of the box. I am using Flowroute as my provider right now and the have confirmed that they utilize T38 in both directions. Thanks! Robert _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130117/f5bce5cc/attachment-0001.html From gvvsubhashkumar at gmail.com Fri Jan 18 09:27:44 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Fri, 18 Jan 2013 11:57:44 +0530 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: Yes,we are trying to bridge PSTN and VOIP call. We posted the log in pastebin.Please use the below URL to access the freeswitch log. http://pastebin.freeswitch.org/20461 Thanks, Subhash. On Thu, Jan 17, 2013 at 10:25 PM, Moises Silva wrote: > On Thu, Jan 17, 2013 at 4:41 AM, Subhash wrote: > >> Hi Michael, >> >> Below is the snippet of the freeswitch log. >> >> Success Call : >> >> 2013-01-16 00:58:58.314163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 >> 2013-01-16 00:58:58.314163 [DEBUG] ftdm_io.c:3038 [s4c1][4:1] Created >> DTMF buffer >> 2013-01-16 00:58:58.314163 [DEBUG] switch_ivr_bridge.c:393 Send signal >> FreeTDM/4:1/2001 [BREAK] >> 2013-01-16 00:58:58.314163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >> DTMF [9] >> 2013-01-16 00:59:06.414163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 1:640 >> 2013-01-16 00:59:06.414163 [DEBUG] switch_ivr_bridge.c:393 Send signal >> FreeTDM/4:1/2001 [BREAK] >> 2013-01-16 00:59:06.414163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >> DTMF [1] >> 2013-01-16 00:59:06.774163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 >> 2013-01-16 00:59:06.774163 [DEBUG] switch_ivr_bridge.c:393 Send signal >> FreeTDM/4:1/2001 [BREAK] >> 2013-01-16 00:59:06.774163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >> DTMF [9] >> 2013-01-16 00:59:09.214164 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read >> wanpipe event 3 >> 2013-01-16 00:59:09.254162 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read >> wanpipe event 3 >> 2013-01-16 00:59:09.254162 [DEBUG] ftmod_wanpipe.c:1547 [s4c1][4:1] >> Queuing wanpipe DTMF: 7 >> 2013-01-16 00:59:09.254162 [DEBUG] ftdm_io.c:3783 [s4c1][4:1] Queuing >> DTMF 7 (debug = 0) >> 2013-01-16 00:59:09.254162 [DEBUG] mod_freetdm.c:811 Queuing DTMF [7] in >> channel FreeTDM/4:1/2001 device 4:1 >> 2013-01-16 00:59:09.254162 [DEBUG] switch_ivr_bridge.c:393 Send signal >> sofia/VOIPBRD1/2001@**.**.**.** [BREAK] >> 2013-01-16 00:59:09.514163 [DEBUG] switch_rtp.c:2589 Queue digit delay of >> 40ms >> 2013-01-16 00:59:09.594164 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read >> wanpipe event 3 >> 2013-01-16 00:59:09.654166 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read >> wanpipe event 3 >> 2013-01-16 00:59:09.654166 [DEBUG] ftmod_wanpipe.c:1547 [s4c1][4:1] >> Queuing wanpipe DTMF: 1 >> 2013-01-16 00:59:09.654166 [DEBUG] ftdm_io.c:3783 [s4c1][4:1] Queuing >> DTMF 1 (debug = 0) >> 2013-01-16 00:59:09.674165 [DEBUG] mod_freetdm.c:811 Queuing DTMF [1] in >> channel FreeTDM/4:1/2001 device 4:1 >> 2013-01-16 00:59:09.674165 [DEBUG] switch_ivr_bridge.c:393 Send signal >> sofia/VOIPBRD1/2001@**.**.**.** [BREAK] >> 013-01-16 00:59:09.934163 [DEBUG] switch_rtp.c:2589 Queue digit delay of >> 40ms >> 2013-01-16 00:59:11.234163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 >> 2013-01-16 00:59:11.234163 [DEBUG] switch_ivr_bridge.c:393 Send signal >> FreeTDM/4:1/2001 [BREAK] >> 2013-01-16 00:59:11.234163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >> DTMF [9] >> 2013-01-16 00:59:11.394163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 >> 2013-01-16 00:59:11.394163 [DEBUG] switch_ivr_bridge.c:393 Send signal >> FreeTDM/4:1/2001 [BREAK] >> 2013-01-16 00:59:11.394163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >> DTMF [9] >> Failed DTMF call : >> >> 2013-01-16 01:02:27.694163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 >> 2013-01-16 01:02:27.694163 [DEBUG] switch_ivr_bridge.c:393 Send signal >> FreeTDM/4:1/2001 [BREAK] >> 2013-01-16 01:02:27.694163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >> DTMF [9] >> 2013-01-16 01:02:32.834164 [INFO] ftmod_sangoma_isdn_stack_rcv.c:232 >> [s4c1][4:1] Received DISCONNECT (suId:1 suInstId:2 spInstId:2) >> 2013-01-16 01:02:32.834164 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:543 >> [s4c1][4:1] Processing DISCONNECT (suId:1 suInstId:2 >> >> > This looks like incoming DTMF from the RTP leg going out in the TDM leg? > if so, the problem would be why switch_rtp.c is "detecting" received DTMF > when there is none? > > Please post in pastebin the full debug log (from call start, bridge, to > hangup etc) for one success call followed by the call with the dtmf problem. > > > *Moises Silva > **Manager, Software Engineering*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > ** > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter`| > | YouTube > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/2c718a32/attachment.html From admin at smallunix.net Fri Jan 18 11:01:28 2013 From: admin at smallunix.net (Andrea Mazzeo) Date: Fri, 18 Jan 2013 09:01:28 +0100 Subject: [Freeswitch-users] SONUS - dtg inside INVITE In-Reply-To: <5D31EF90-20B7-48BA-B7BC-65F30B5383DD@jerris.com> References: <7443520999689776200@unknownmsgid> <5D31EF90-20B7-48BA-B7BC-65F30B5383DD@jerris.com> Message-ID: Hi Michael, In the contact field. 2013/1/18 Michael Jerris : > What you posted is NOT a valid invite header. WHERE in the invite header do you need it? > > On Jan 17, 2013, at 6:59 PM, Andrea Mazzeo wrote: > >> Hi Kristian, >> >> I need dtg in the INVITE header. >> >> Best regards, >> Andrea Mazzeo >> >> Il giorno 17/gen/2013, alle ore 22:01, Kristian Kielhofner >> ha scritto: >> >>> This is almost certainly not the syntax you want to use. >>> >>> Do that expect "dtg" to be a user or domain parameter? >>> >>> On Thu, Jan 17, 2013 at 1:46 PM, Andrea Mazzeo wrote: >>>> Hi all, >>>> >>>> I need to configure my FS to send call to a Sonus GSX via SIP. >>>> Is there a way to send SIP parameter "dtg" (destination trunk group) >>>> inside the INVITE? >>>> >>>> Example: >>>> >>>> Request-Line: INVITE sip: 39340xxxxxx at 38.xxx.xxx.xxx SIP/2.0 dtg=ATTMDT7807IPO >>>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vbvbrj at gmail.com Fri Jan 18 11:37:01 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 18 Jan 2013 10:37:01 +0200 Subject: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 Message-ID: <50F909AD.2010308@gmail.com> Hello. Randomly I get this in kern.log: kernel: freeswitch[3136]: segfault at 8d00000081 ip 00007fbe75c90130 sp 00007fbe705c66d0 error 4 in libfreeswitch.so.1.0.0[7fbe75b68000+277000] Its so random that I can't find why. Usually this happen once a day at random time. There are days when this does not happen. echo 1 > /proc/sys/fs/suid_dumpable ulimit -c unlimited As per wiki, core files can not be located. Ulimit is enabled before starting freeswitch. But as FS is started under user from basedir directory, and that user does not have write permission to this directory, therefore no core files are created. The FS version is: FreeSWITCH Version 1.3.8b+git~20121130T003427Z~b35ac6b3bd (git b35ac6b 2012-11-30 00:34:27Z) Maybe something was changed from that version. In jira changelog I didn't see any related to this. How to track this? Is there a variable to set for core files directory dump? Thank you. -- Mimiko desu. From vbvbrj at gmail.com Fri Jan 18 12:23:28 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 18 Jan 2013 11:23:28 +0200 Subject: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 In-Reply-To: <50F909AD.2010308@gmail.com> References: <50F909AD.2010308@gmail.com> Message-ID: <50F91490.7080006@gmail.com> New one: kernel: freeswitch[3559]: segfault at 30 ip 00007fbb945862a4 sp 00007fbb8ef59bc0 error 4 in libfreeswitch.so.1.0.0[7fbb944fb000+277000] This happened when channels where destroyed after a bridge and a hangup. -- Mimiko desu. From peter.olsson at visionutveckling.se Fri Jan 18 12:33:04 2013 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 18 Jan 2013 09:33:04 +0000 Subject: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 Message-ID: <1FFF97C269757C458224B7C895F35F15205F66@cantor.std.visionutv.se> First of all - try latest from git head. If it still exists, report to Jira. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Mimiko Skickat: den 18 januari 2013 10:23 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 New one: kernel: freeswitch[3559]: segfault at 30 ip 00007fbb945862a4 sp 00007fbb8ef59bc0 error 4 in libfreeswitch.so.1.0.0[7fbb944fb000+277000] This happened when channels where destroyed after a bridge and a hangup. -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:50f911b532761664217464! From paul at cupis.co.uk Fri Jan 18 13:26:14 2013 From: paul at cupis.co.uk (Paul Cupis) Date: Fri, 18 Jan 2013 10:26:14 +0000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> Message-ID: <20130118102614.GA27116@eagle.cupis.co.uk> On Fri, Jan 18, 2013 at 01:57:36AM +0000, Steven Ayre wrote: > > > What're you hoping to do here? This sets the payload type value. 0 is a > valid pt. This won't disable it, it'll instead use the value 0. That > happens to be the one for G711, so if that codec is enabled you're just > going to get a conflict. If the rfc2833-pt is set to a value <96 it is outside the allowed range for dynamic payload negotiation and FS will not offer RFC2833 support in the SDP. ref: sofia_glue.c Regards, From gassaad at emassembly.com Fri Jan 18 15:49:46 2013 From: gassaad at emassembly.com (George Assaad) Date: Fri, 18 Jan 2013 07:49:46 -0500 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: Message-ID: Although the gui mentioned to attach a tiff, I finds it to not work. Please try pdf, you should be in business now. Please let us know, On Jan 18, 2013 12:35 AM, "Robert Craig" wrote: > OK, so I've made a little bit of progress. The distant end is a Windows > Fax server hooked up to an analog line. I created a fax extension in FS, > assigned an inbound route to the extension, and made sure the extension had > an email address. I believe the fax receive wants to complete, but there is > an interesting error in the log about TIF files. Anyone ever seen this > before? > > v=0 > o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > s=FreeSWITCH > c=IN IP4 72.201.112.131 > t=0 0 > m=image 27714 udptl t38 > a=T38FaxVersion:0 > a=T38MaxBitRate:9600 > a=T38FaxFillBitRemoval > a=T38FaxRateManagement:transferredTCF > a=T38FaxMaxBuffer:2000 > a=T38FaxMaxDatagram:400 > a=T38FaxUdpEC:t38UDPRedundancy > > 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct ip/port > confirmed. > 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === Negotiation > Result ======================================================= > 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote station > id: ATW-4792460547 > 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local station id: > SpanDSP Fax Ident > 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer Rate: > 9600 > 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM status off > 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote country: > 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote vendor: > 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote model: > 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > ============================================================================== > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > ============================================================================== > *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax processing > not successful - result (41) TIFF/F file cannot be opened.* > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote station > id: ATW-4792460547 > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local station id: > SpanDSP Fax Ident > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > transferred: 0 > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: > 0 > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image resolution: > 0x0 > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: > 9600 > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM status off > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote country: > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote vendor: > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote model: > 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > ============================================================================== > 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 sofia/external/ > +14792460547 at flowroute.com Restore previous codec PCMU:0. > EXECUTE sofia/external/+14792460547 at flowroute.com hangup() > > Robert > > > ------------------------------ > From: craig.robert at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Wed, 16 Jan 2013 10:40:31 -0700 > Subject: Re: [Freeswitch-users] T38 Configuration > > OK, I am not extremely concerned with getting an ATA working right now. If > I could just get an inbound extension-to-email working it would be great. > I'll start from scratch and then let you know what happens. Just to make > sure, I am going to create an extension, assign an email address to it, > make sure there is an inbound route to that extension, and then test it. > > > Robert > > ------------------------------ > Date: Wed, 16 Jan 2013 12:11:34 -0500 > From: gassaad at emassembly.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > The reason for outbound extension is: > When you send a fax from ATA to the outside world, it will dial first to > the outbound extension and then after successful receive it will dial out > to the intended number. > > As for the error you have it may be related to your ATA. I am not sure > please see why do you get the following message. > "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > > On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig wrote: > > OK, I've already tried creating one extension for inbound and set an email > address. Of course, it failed and the errors are at the bottom of this > email. What is the purpose of the outbound extension? > > Robert > > > ------------------------------ > Date: Wed, 16 Jan 2013 09:57:51 -0500 > From: gassaad at emassembly.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Sorry for the delay to get back to you. > 1- App --> fax server > 2- Create two extensions (one for inbound) another for (outbound) From > within either extension you can send faxes "virtual fax" > 3- point your create your outbound and inbound dialplans as normal but > make it to point to those extensions. > > Good luck, > > George > > On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig wrote: > > What did you do on the config side to get it to work? I've tried both an > Ata and a fax to email extension and they both fail in both directions. > > Robert > > On Jan 12, 2013, at 2:04 PM, "George Assaad" > wrote: > > I use Fusion successfully to send / receive fax. > The way it works: > ATA --->FaxServer. > After successful receipt, Fax Server---> DID where you > have dialled originally. > > Cheers, > > George > > > On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig wrote: > > Did anyone see the issue or what I may have missed in the config? As > mentioned earlier, I thought it would after a fresh install. Any help is > appreciated. > > Robert > > > ------------------------------ > From: craig.robert at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Thu, 10 Jan 2013 13:43:09 -0700 > > Subject: Re: [Freeswitch-users] T38 Configuration > > Thanks for the quick response. Below is the capture from the log viewer. > In this test, I am using a remote fax machine (line from Cox with physical > machine) faxing into the "email server" where I created an extension for > fax-to-email. Calling number is 1479-246-0547. > > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 > (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 > sofia/external/+14792460547 at flowroute.com Standard DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/ > +14792460547 at flowroute.com SOFIA DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 > (sofia/external/+14792460547 at flowroute.com) State DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 > (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> > DOWN > 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close > Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] > 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 > (sofia/external/+14792460547 at flowroute.com) Ended > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 > (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external > entities > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> > CS_DESTROY > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 > (sofia/external/+14792460547 at flowroute.com) State REPORTING going to > sleep > 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 > sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: > NORMAL_CLEARING > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 > (sofia/external/+14792460547 at flowroute.com) State REPORTING > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_REPORTING > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 > (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> > CS_REPORTING > > 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup > Command with no Session system(/usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 > name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after > permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 > caller_id_name='' caller_id_number=): > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 > (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 > sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: > NORMAL_CLEARING > 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to > sofia/external/+14792460547 at flowroute.com > 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel sofia/external/+ > 14792460547 at flowroute.com hanging up, cause: NORMAL_CLEARING > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 > (sofia/external/+14792460547 at flowroute.com) State HANGUP > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_HANGUP > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 > sofia/external/+14792460547 at flowroute.com skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal > sofia/external/+14792460547 at flowroute.com [KILL] > 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup > sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] > 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+ > 14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP > EXECUTE sofia/external/+14792460547 at flowroute.com hangup() > 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 sofia/external/+ > 14792460547 at flowroute.com Restore previous codec PCMU:0. > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 > ============================================================================== > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: > 14400 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image resolution: > 0x0 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: > 0 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages > transferred: 0 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station id: > SpanDSP Fax Ident > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station > id: ATW-4792460547 > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing > not successful - result (48) Disconnected after permitted retries. > 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 > ============================================================================== > 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port > confirmed. > 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+ > 14792460547 at flowroute.com image media sdp: > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+ > 14792460547 at flowroute.com image media sdp: > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ > 66.42.121.5:31632] has not changed. > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+ > 14792460547 at flowroute.com Set rtp dtmf delay to 40 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive > payload to 101 > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send > payload to 101 > 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer > spec [0] must be between 20 and 10000 > 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] > 160 bytes per 20ms > 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP > [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> > 66.42.121.5 port 31632 codec: 0 ms: 20 > 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [ready][200] > 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxRateManagement:transferredTCF > m=image 31632 udptl t38 > t=0 0 > c=IN IP4 66.42.121.5 > s=- > o=- 275426305 1357849876 IN IP4 66.42.121.5 > v=0 > 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: > 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [completing][200] > 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+ > 14792460547 at flowroute.com Update Callee ID to "+14792460547" <+14792460547 > > > 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [calling][0] > 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: > > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxMaxDatagram:400 > a=T38FaxMaxBuffer:2000 > a=T38FaxRateManagement:transferredTCF > a=T38FaxFillBitRemoval > a=T38MaxBitRate:9600 > a=T38FaxVersion:0 > m=image 27588 udptl t38 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+ > 14792460547 at flowroute.com image media sdp: > 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec > activation Success L16 > 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 sofia/external/+ > 14792460547 at flowroute.com Push codec L16:70 > 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec > activation Success L16 20000 > EXECUTE sofia/external/+14792460547 at flowroute.comrxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done playing > file silence_stream://2000 > 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port > confirmed. > 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [ready][200] > 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec > Activated L16 at 8000hz 1 channels 20ms > EXECUTE sofia/external/+14792460547 at flowroute.complayback(silence_stream://2000) > 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET > [last_fax]=[+14792460547-2013-01-10-13-39-26] > EXECUTE sofia/external/+14792460547 at flowroute.comset(last_fax=+14792460547-2013-01-10-13-39-26) > 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] > EXECUTE sofia/external/+14792460547 at flowroute.comset(fax_enable_t38_request=true) > 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [fax_enable_t38]=[true] > 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [completed][200] > EXECUTE sofia/external/+14792460547 at flowroute.comset(fax_enable_t38=true) > 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel > [sofia/external/+14792460547 at flowroute.com] has been answered > 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+ > 14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE > 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > > a=sendrecv > a=ptime:20 > a=silenceSupp:off - - - - > a=fmtp:101 0-16 > a=rtpmap:101 telephone-event/8000 > a=rtpmap:0 PCMU/8000 > m=audio 27588 RTP/AVP 0 101 > t=0 0 > c=IN IP4 72.201.112.131 > s=FreeSWITCH > o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 > v=0 > 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP > sofia/external/+14792460547 at flowroute.com: > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+ > 14792460547 at flowroute.com Set rtp dtmf delay to 40 > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive > payload to 101 > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send > payload to 101 > 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] > 160 bytes per 20ms > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP > [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> > 66.42.121.5 port 31632 codec: 0 ms: 20 > 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success > [72.201.112.131]:[27588] > EXECUTE sofia/external/+14792460547 at flowroute.com answer() > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 name=${last_fax} > messages='result: ${fax_result_text} sender:${fax_remote_station_id} > pages:${fax_document_total_pages}' domain=192.168.4.15 > caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] > EXECUTE sofia/external/+14792460547 at flowroute.comset(api_hangup_hook=system /usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} > messages='result: \${fax_result_text} sender:\${fax_remote_station_id} > pages:\${fax_document_total_pages}' domain=192.168.4.15 > caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 > -0700] > EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, > 10 Jan 2013 13:39:25 -0700) > EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-last_dial/+14792460547/3150) > EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-spymap/ > +14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [call_direction]=[local] > EXECUTE sofia/external/+14792460547 at flowroute.comset(call_direction=local) > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 > sofia/external/+14792460547 at flowroute.com Standard EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+ > 14792460547 at flowroute.com SOFIA EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 > (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> > CS_EXECUTE > Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() > Dialplan: sofia/external/+14792460547 at flowroute.com Action > rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > playback(silence_stream://2000) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(fax_enable_t38_request=true) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(fax_enable_t38=true) > Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(api_hangup_hook=system /usr/bin/php > /var/www/html/fusionpbx/secure/fax_to_email.php email= > robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} > messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} > pages:\\\${fax_document_total_pages}' domain=192.168.4.15 > caller_id_name='\\\${caller_id_name}' > caller_id_number=\\\${caller_id_number}) > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test > Fax] destination_number(3150) =~ /^3150$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->Test Fax] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->vmain_user] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] > destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->vmain] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->send_to_voicemail] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->extension-intercom] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->intercept-ext] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->call_return] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->call_privacy] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition > [global] > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->global] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [redial] > destination_number(3150) =~ /^(redial|\*870)$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->redial] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->group-intercept] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action > set(call_direction=local) > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->call-direction] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [default->Flowroute.11d] continue=false > 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing > +14792460547 <+14792460547>->3150 in context default > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 > sofia/external/+14792460547 at flowroute.com Standard ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+ > 14792460547 at flowroute.com SOFIA ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep > 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer > sofia/external/+14792460547 at flowroute.com to XML[3150 at default] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+ > 14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING > EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML > default) > 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT > (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > EXECUTE sofia/external/+14792460547 at flowroute.comexport(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ > 14792460547 at flowroute.com SET [outside_call]=[true] > EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 > sofia/external/+14792460547 at flowroute.com Standard EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+ > 14792460547 at flowroute.com SOFIA EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/+14792460547 at flowroute.com) State EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_EXECUTE > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 > (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> > CS_EXECUTE > Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 > XML default) > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ > break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (PASS) [Test_Number] context(public) =~ /public/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->Test_Number] continue=false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex > (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->call_debug] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/external/+14792460547 at flowroute.com Action > set(outside_call=true) > Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition > [outside_call] > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->outside_call] continue=true > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/+14792460547 at flowroute.com parsing > [public->unloop] continue=false > 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing > +14792460547 <+14792460547>->15203039810 in context public > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 > sofia/external/+14792460547 at flowroute.com Standard ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+ > 14792460547 at flowroute.com SOFIA ROUTING > 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+14792460547 at flowroute.com) State ROUTING > 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+ > 14792460547 at flowroute.com) Callstate Change DOWN -> RINGING > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change > CS_ROUTING > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 > (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+ > 14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING > 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+ > 14792460547 at flowroute.com SOFIA INIT > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 > (sofia/external/+14792460547 at flowroute.com) State INIT > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+ > 14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf > send/recv payload to 101 > 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 sofia/external/+ > 14792460547 at flowroute.com Original read codec set to PCMU:0 > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec > sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples > 64000 bits > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > > a=ptime:20 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=fmtp:18 annexb=no > a=rtpmap:18 G729/8000 > m=audio 31632 RTP/AVP 0 18 101 > t=0 0 > c=IN IP4 66.42.121.5 > s=- > o=- 275426305 1357849875 IN IP4 66.42.121.5 > v=0 > 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: > 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+ > 14792460547 at flowroute.com entering state [received][100] > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 > (sofia/external/+14792460547 at flowroute.com) State NEW > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/+14792460547 at flowroute.com [BREAK] > 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel > sofia/external/+14792460547 at flowroute.com[d276f012-5b65-11e2-beae-8564ff827a25] > > Robert > > > ------------------------------ > Date: Thu, 10 Jan 2013 09:47:18 -0800 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Can you pastebin a console debug log w/ sip trace of a call failing to > negotiate t.38 correctly? That might help us figure out where to look. > -MC > > On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: > > Just installed FusionPBX using the script built for a Centos 6 install. > Everything works great in regards to calls, but my main goal of this > project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have > tried many things in media bypass, routing to a fax extension, using an > ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me > in the right direction as to where I should begin? I am guessing it isn't > supposed to work right out of the box. I am using Flowroute as my provider > right now and the have confirmed that they utilize T38 in both directions. > Thanks! > > > Robert > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/286bb878/attachment-0001.html From craig.robert at hotmail.com Fri Jan 18 16:37:29 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Fri, 18 Jan 2013 06:37:29 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: Message-ID: This was an incoming fax. I don't have much control over the other end. Why is that error appearing when trying to receive something? Robert On Jan 18, 2013, at 5:58 AM, "George Assaad" wrote: > Although the gui mentioned to attach a tiff, I finds it to not work. Please try pdf, you should be in business now. > Please let us know, > > On Jan 18, 2013 12:35 AM, "Robert Craig" wrote: >> OK, so I've made a little bit of progress. The distant end is a Windows Fax server hooked up to an analog line. I created a fax extension in FS, assigned an inbound route to the extension, and made sure the extension had an email address. I believe the fax receive wants to complete, but there is an interesting error in the log about TIF files. Anyone ever seen this before? >> >> v=0 >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 >> s=FreeSWITCH >> c=IN IP4 72.201.112.131 >> t=0 0 >> m=image 27714 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === Negotiation Result ======================================================= >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote station id: ATW-4792460547 >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local station id: SpanDSP Fax Ident >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer Rate: 9600 >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM status off >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote country: >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote vendor: >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote model: >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 ============================================================================== >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (41) TIFF/F file cannot be opened. >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote station id: ATW-4792460547 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 9600 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM status off >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote country: >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote model: >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. >> EXECUTE sofia/external/+14792460547 at flowroute.com hangup() >> >> Robert >> >> >> >> From: craig.robert at hotmail.com >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 16 Jan 2013 10:40:31 -0700 >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> OK, I am not extremely concerned with getting an ATA working right now. If I could just get an inbound extension-to-email working it would be great. I'll start from scratch and then let you know what happens. Just to make sure, I am going to create an extension, assign an email address to it, make sure there is an inbound route to that extension, and then test it. >> >> >> Robert >> >> >> Date: Wed, 16 Jan 2013 12:11:34 -0500 >> From: gassaad at emassembly.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> The reason for outbound extension is: >> When you send a fax from ATA to the outside world, it will dial first to the outbound extension and then after successful receive it will dial out to the intended number. >> >> As for the error you have it may be related to your ATA. I am not sure please see why do you get the following message. >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " >> >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig wrote: >> OK, I've already tried creating one extension for inbound and set an email address. Of course, it failed and the errors are at the bottom of this email. What is the purpose of the outbound extension? >> >> Robert >> >> >> Date: Wed, 16 Jan 2013 09:57:51 -0500 >> From: gassaad at emassembly.com >> >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Sorry for the delay to get back to you. >> 1- App --> fax server >> 2- Create two extensions (one for inbound) another for (outbound) From within either extension you can send faxes "virtual fax" >> 3- point your create your outbound and inbound dialplans as normal but make it to point to those extensions. >> >> Good luck, >> >> George >> >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig wrote: >> What did you do on the config side to get it to work? I've tried both an Ata and a fax to email extension and they both fail in both directions. >> >> Robert >> >> On Jan 12, 2013, at 2:04 PM, "George Assaad" wrote: >> >> I use Fusion successfully to send / receive fax. >> The way it works: >> ATA --->FaxServer. >> After successful receipt, Fax Server---> DID where you have dialled originally. >> >> Cheers, >> >> George >> >> >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig wrote: >> Did anyone see the issue or what I may have missed in the config? As mentioned earlier, I thought it would after a fresh install. Any help is appreciated. >> >> Robert >> >> >> From: craig.robert at hotmail.com >> To: freeswitch-users at lists.freeswitch.org >> Date: Thu, 10 Jan 2013 13:43:09 -0700 >> >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Thanks for the quick response. Below is the capture from the log viewer. In this test, I am using a remote fax machine (line from Cox with physical machine) faxing into the "email server" where I created an extension for fax-to-email. Calling number is 1479-246-0547. >> >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 sofia/external/+14792460547 at flowroute.com Standard DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/+14792460547 at flowroute.com SOFIA DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> DOWN >> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] >> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 (sofia/external/+14792460547 at flowroute.com) Ended >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external entities >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING going to sleep >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: NORMAL_CLEARING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> CS_REPORTING >> >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup Command with no Session system(/usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 caller_id_name='' caller_id_number=): >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to sofia/external/+14792460547 at flowroute.com >> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel sofia/external/+14792460547 at flowroute.com hanging up, cause: NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 sofia/external/+14792460547 at flowroute.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal sofia/external/+14792460547 at flowroute.com [KILL] >> 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] >> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP >> EXECUTE sofia/external/+14792460547 at flowroute.com hangup() >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 14400 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station id: ATW-4792460547 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (48) Disconnected after permitted retries. >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== >> 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. >> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 >> 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer spec [0] must be between 20 and 10000 >> 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] >> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxRateManagement:transferredTCF >> m=image 31632 udptl t38 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849876 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completing][200] >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+14792460547 at flowroute.com Update Callee ID to "+14792460547" <+14792460547> >> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [calling][0] >> 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec activation Success L16 >> 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 sofia/external/+14792460547 at flowroute.com Push codec L16:70 >> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec activation Success L16 20000 >> EXECUTE sofia/external/+14792460547 at flowroute.com rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) >> 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done playing file silence_stream://2000 >> 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms >> EXECUTE sofia/external/+14792460547 at flowroute.com playback(silence_stream://2000) >> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [last_fax]=[+14792460547-2013-01-10-13-39-26] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(last_fax=+14792460547-2013-01-10-13-39-26) >> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38_request=true) >> 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38]=[true] >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completed][200] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38=true) >> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel [sofia/external/+14792460547 at flowroute.com] has been answered >> 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE >> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=sendrecv >> a=ptime:20 >> a=silenceSupp:off - - - - >> a=fmtp:101 0-16 >> a=rtpmap:101 telephone-event/8000 >> a=rtpmap:0 PCMU/8000 >> m=audio 27588 RTP/AVP 0 101 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP sofia/external/+14792460547 at flowroute.com: >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success [72.201.112.131]:[27588] >> EXECUTE sofia/external/+14792460547 at flowroute.com answer() >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=${last_fax} messages='result: ${fax_result_text} sender:${fax_remote_station_id} pages:${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} messages='result: \${fax_result_text} sender:\${fax_remote_station_id} pages:\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) >> EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) >> EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/+14792460547/3150) >> EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [call_direction]=[local] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(call_direction=local) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE >> Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() >> Dialplan: sofia/external/+14792460547 at flowroute.com Action rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action playback(silence_stream://2000) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38_request=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} pages:\\\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\\\${caller_id_name}' caller_id_number=\\\${caller_id_number}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test Fax] destination_number(3150) =~ /^3150$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Test Fax] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain_user] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->send_to_voicemail] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->extension-intercom] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->intercept-ext] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_return] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_privacy] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/global/${uuid}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [global] >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->global] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [redial] destination_number(3150) =~ /^(redial|\*870)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->redial] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->group-intercept] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action set(call_direction=local) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call-direction] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Flowroute.11d] continue=false >> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->3150 in context default >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep >> 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer sofia/external/+14792460547 at flowroute.com to XML[3150 at default] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING >> EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML default) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.com export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [outside_call]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE >> Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 XML default) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] context(public) =~ /public/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->Test_Number] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->call_debug] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(outside_call=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [outside_call] >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->outside_call] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->unloop] continue=false >> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->15203039810 in context public >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+14792460547 at flowroute.com) Callstate Change DOWN -> RINGING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+14792460547 at flowroute.com SOFIA INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf send/recv payload to 101 >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 sofia/external/+14792460547 at flowroute.com Original read codec set to PCMU:0 >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples 64000 bits >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >> >> a=ptime:20 >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:18 G729/8000 >> m=audio 31632 RTP/AVP 0 18 101 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849875 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [received][100] >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 (sofia/external/+14792460547 at flowroute.com) State NEW >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel sofia/external/+14792460547 at flowroute.com [d276f012-5b65-11e2-beae-8564ff827a25] >> >> Robert >> >> >> Date: Thu, 10 Jan 2013 09:47:18 -0800 >> From: msc at freeswitch.org >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Can you pastebin a console debug log w/ sip trace of a call failing to negotiate t.38 correctly? That might help us figure out where to look. >> -MC >> >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: >> Just installed FusionPBX using the script built for a Centos 6 install. Everything works great in regards to calls, but my main goal of this project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have tried many things in media bypass, routing to a fax extension, using an ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me in the right direction as to where I should begin? I am guessing it isn't supposed to work right out of the box. I am using Flowroute as my provider right now and the have confirmed that they utilize T38 in both directions. Thanks! >> >> >> Robert >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> >> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> >> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/84cddd48/attachment-0001.html -------------- next part -------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gassaad at emassembly.com Fri Jan 18 17:04:07 2013 From: gassaad at emassembly.com (George Assaad) Date: Fri, 18 Jan 2013 09:04:07 -0500 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: Message-ID: Will check mine tonight On Jan 18, 2013 8:41 AM, "Robert Craig" wrote: > This was an incoming fax. I don't have much control over the other end. > Why is that error appearing when trying to receive something? > > Robert > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > wrote: > > Although the gui mentioned to attach a tiff, I finds it to not work. > Please try pdf, you should be in business now. > Please let us know, > On Jan 18, 2013 12:35 AM, "Robert Craig" wrote: > >> OK, so I've made a little bit of progress. The distant end is a Windows >> Fax server hooked up to an analog line. I created a fax extension in FS, >> assigned an inbound route to the extension, and made sure the extension had >> an email address. I believe the fax receive wants to complete, but there is >> an interesting error in the log about TIF files. Anyone ever seen this >> before? >> >> v=0 >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 >> s=FreeSWITCH >> c=IN IP4 72.201.112.131 >> t=0 0 >> m=image 27714 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct ip/port >> confirmed. >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === Negotiation >> Result ======================================================= >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote station >> id: ATW-4792460547 >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local station >> id: SpanDSP Fax Ident >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer Rate: >> 9600 >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM status off >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote country: >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote vendor: >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote model: >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 >> ============================================================================== >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 >> ============================================================================== >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax processing >> not successful - result (41) TIFF/F file cannot be opened.* >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote station >> id: ATW-4792460547 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local station >> id: SpanDSP Fax Ident >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages >> transferred: 0 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: >> 0 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image >> resolution: 0x0 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: >> 9600 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM status off >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote country: >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote model: >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 >> ============================================================================== >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 sofia/external/ >> +14792460547 at flowroute.com Restore previous codec PCMU:0. >> EXECUTE sofia/external/+14792460547 at flowroute.com hangup() >> >> Robert >> >> >> ------------------------------ >> From: craig.robert at hotmail.com >> To: freeswitch-users at lists.freeswitch.org >> Date: Wed, 16 Jan 2013 10:40:31 -0700 >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> OK, I am not extremely concerned with getting an ATA working right now. >> If I could just get an inbound extension-to-email working it would be >> great. I'll start from scratch and then let you know what happens. Just to >> make sure, I am going to create an extension, assign an email address to >> it, make sure there is an inbound route to that extension, and then test it. >> >> >> Robert >> >> ------------------------------ >> Date: Wed, 16 Jan 2013 12:11:34 -0500 >> From: gassaad at emassembly.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> The reason for outbound extension is: >> When you send a fax from ATA to the outside world, it will dial first to >> the outbound extension and then after successful receive it will dial out >> to the intended number. >> >> As for the error you have it may be related to your ATA. I am not sure >> please see why do you get the following message. >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " >> >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig wrote: >> >> OK, I've already tried creating one extension for inbound and set an >> email address. Of course, it failed and the errors are at the bottom of >> this email. What is the purpose of the outbound extension? >> >> Robert >> >> >> ------------------------------ >> Date: Wed, 16 Jan 2013 09:57:51 -0500 >> From: gassaad at emassembly.com >> >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Sorry for the delay to get back to you. >> 1- App --> fax server >> 2- Create two extensions (one for inbound) another for (outbound) From >> within either extension you can send faxes "virtual fax" >> 3- point your create your outbound and inbound dialplans as normal but >> make it to point to those extensions. >> >> Good luck, >> >> George >> >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig wrote: >> >> What did you do on the config side to get it to work? I've tried both an >> Ata and a fax to email extension and they both fail in both directions. >> >> Robert >> >> On Jan 12, 2013, at 2:04 PM, "George Assaad" >> wrote: >> >> I use Fusion successfully to send / receive fax. >> The way it works: >> ATA --->FaxServer. >> After successful receipt, Fax Server---> DID where you >> have dialled originally. >> >> Cheers, >> >> George >> >> >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig wrote: >> >> Did anyone see the issue or what I may have missed in the config? As >> mentioned earlier, I thought it would after a fresh install. Any help is >> appreciated. >> >> Robert >> >> >> ------------------------------ >> From: craig.robert at hotmail.com >> To: freeswitch-users at lists.freeswitch.org >> Date: Thu, 10 Jan 2013 13:43:09 -0700 >> >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Thanks for the quick response. Below is the capture from the log viewer. >> In this test, I am using a remote fax machine (line from Cox with physical >> machine) faxing into the "email server" where I created an extension for >> fax-to-email. Calling number is 1479-246-0547. >> >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 >> (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 >> sofia/external/+14792460547 at flowroute.com Standard DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/ >> +14792460547 at flowroute.com SOFIA DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 >> (sofia/external/+14792460547 at flowroute.com) State DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 >> (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> >> DOWN >> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close >> Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] >> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 >> (sofia/external/+14792460547 at flowroute.com) Ended >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 >> (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external >> entities >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 >> (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> >> CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 >> (sofia/external/+14792460547 at flowroute.com) State REPORTING going to >> sleep >> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 >> sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: >> NORMAL_CLEARING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 >> (sofia/external/+14792460547 at flowroute.com) State REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 >> (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> >> CS_REPORTING >> >> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup >> Command with no Session system(/usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php email= >> robert.l.craig at craignetwork.com extension=3150 >> name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after >> permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 >> caller_id_name='' caller_id_number=): >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 >> (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 >> sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: >> NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to >> sofia/external/+14792460547 at flowroute.com >> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel >> sofia/external/+14792460547 at flowroute.com hanging up, cause: >> NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 >> (sofia/external/+14792460547 at flowroute.com) State HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 >> sofia/external/+14792460547 at flowroute.com skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal >> sofia/external/+14792460547 at flowroute.com [KILL] >> 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup >> sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] >> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+ >> 14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP >> EXECUTE sofia/external/+14792460547 at flowroute.com hangup() >> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 >> sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 >> ============================================================================== >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: >> 14400 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image >> resolution: 0x0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: >> 0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages >> transferred: 0 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station >> id: SpanDSP Fax Ident >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station >> id: ATW-4792460547 >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing >> not successful - result (48) Disconnected after permitted retries. >> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 >> ============================================================================== >> 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port >> confirmed. >> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+ >> 14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ >> 66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ >> 66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+ >> 14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ >> 66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [ >> 66.42.121.5:31632] has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+ >> 14792460547 at flowroute.com Set rtp dtmf delay to 40 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf >> receive payload to 101 >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send >> payload to 101 >> 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer >> spec [0] must be between 20 and 10000 >> 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer >> [soft] 160 bytes per 20ms >> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP >> [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> >> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [ready][200] >> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxRateManagement:transferredTCF >> m=image 31632 udptl t38 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849876 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [completing][200] >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+ >> 14792460547 at flowroute.com Update Callee ID to "+14792460547" < >> +14792460547> >> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [calling][0] >> 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+ >> 14792460547 at flowroute.com image media sdp: >> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec >> activation Success L16 >> 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 >> sofia/external/+14792460547 at flowroute.com Push codec L16:70 >> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec >> activation Success L16 20000 >> EXECUTE sofia/external/+14792460547 at flowroute.comrxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) >> 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done >> playing file silence_stream://2000 >> 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port >> confirmed. >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [ready][200] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec >> Activated L16 at 8000hz 1 channels 20ms >> EXECUTE sofia/external/+14792460547 at flowroute.complayback(silence_stream://2000) >> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET >> [last_fax]=[+14792460547-2013-01-10-13-39-26] >> EXECUTE sofia/external/+14792460547 at flowroute.comset(last_fax=+14792460547-2013-01-10-13-39-26) >> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.comset(fax_enable_t38_request=true) >> 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [fax_enable_t38]=[true] >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [completed][200] >> EXECUTE sofia/external/+14792460547 at flowroute.comset(fax_enable_t38=true) >> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel >> [sofia/external/+14792460547 at flowroute.com] has been answered >> 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+ >> 14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE >> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> >> a=sendrecv >> a=ptime:20 >> a=silenceSupp:off - - - - >> a=fmtp:101 0-16 >> a=rtpmap:101 telephone-event/8000 >> a=rtpmap:0 PCMU/8000 >> m=audio 27588 RTP/AVP 0 101 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 >> v=0 >> 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP >> sofia/external/+14792460547 at flowroute.com: >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+ >> 14792460547 at flowroute.com Set rtp dtmf delay to 40 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf >> receive payload to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send >> payload to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer >> [soft] 160 bytes per 20ms >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP >> [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> >> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success >> [72.201.112.131]:[27588] >> EXECUTE sofia/external/+14792460547 at flowroute.com answer() >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php email= >> robert.l.craig at craignetwork.com extension=3150 name=${last_fax} >> messages='result: ${fax_result_text} sender:${fax_remote_station_id} >> pages:${fax_document_total_pages}' domain=192.168.4.15 >> caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] >> EXECUTE sofia/external/+14792460547 at flowroute.comset(api_hangup_hook=system /usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php email= >> robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} >> messages='result: \${fax_result_text} sender:\${fax_remote_station_id} >> pages:\${fax_document_total_pages}' domain=192.168.4.15 >> caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 >> -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, >> 10 Jan 2013 13:39:25 -0700) >> EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) >> EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-last_dial/+14792460547/3150) >> EXECUTE sofia/external/+14792460547 at flowroute.comhash(insert/192.168.4.15-spymap/ >> +14792460547/d276f012-5b65-11e2-beae-8564ff827a25) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [call_direction]=[local] >> EXECUTE sofia/external/+14792460547 at flowroute.comset(call_direction=local) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 >> sofia/external/+14792460547 at flowroute.com Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+ >> 14792460547 at flowroute.com SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 >> (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> >> CS_EXECUTE >> Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> playback(silence_stream://2000) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(fax_enable_t38_request=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(fax_enable_t38=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(api_hangup_hook=system /usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php email= >> robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} >> messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} >> pages:\\\${fax_document_total_pages}' domain=192.168.4.15 >> caller_id_name='\\\${caller_id_name}' >> caller_id_number=\\\${caller_id_number}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test >> Fax] destination_number(3150) =~ /^3150$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->Test Fax] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->vmain_user] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] >> destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->vmain] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->send_to_voicemail] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->extension-intercom] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->intercept-ext] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->call_return] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->call_privacy] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> hash(insert/${domain_name}-last_dial/global/${uuid}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition >> [global] >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->global] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [redial] destination_number(3150) =~ /^(redial|\*870)$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->redial] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->group-intercept] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action >> set(call_direction=local) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->call-direction] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [default->Flowroute.11d] continue=false >> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing >> +14792460547 <+14792460547>->3150 in context default >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 >> sofia/external/+14792460547 at flowroute.com Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+ >> 14792460547 at flowroute.com SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com) State ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep >> 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer >> sofia/external/+14792460547 at flowroute.com to XML[3150 at default] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+ >> 14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING >> EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML >> default) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT >> (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.comexport(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) >> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+ >> 14792460547 at flowroute.com SET [outside_call]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 >> sofia/external/+14792460547 at flowroute.com Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+ >> 14792460547 at flowroute.com SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 >> (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> >> CS_EXECUTE >> Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 >> XML default) >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ >> break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (PASS) [Test_Number] context(public) =~ /public/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [public->Test_Number] continue=false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [public->call_debug] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/+14792460547 at flowroute.com Action >> set(outside_call=true) >> Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition >> [outside_call] >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [public->outside_call] continue=true >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com Regex >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/+14792460547 at flowroute.com parsing >> [public->unloop] continue=false >> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing >> +14792460547 <+14792460547>->15203039810 in context public >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 >> sofia/external/+14792460547 at flowroute.com Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+ >> 14792460547 at flowroute.com SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com) State ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+ >> 14792460547 at flowroute.com) Callstate Change DOWN -> RINGING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change >> CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 >> (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+ >> 14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+ >> 14792460547 at flowroute.com SOFIA INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 >> (sofia/external/+14792460547 at flowroute.com) State INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+ >> 14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf >> send/recv payload to 101 >> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 >> sofia/external/+14792460547 at flowroute.com Original read codec set to >> PCMU:0 >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec >> sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples >> 64000 bits >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare >> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >> >> a=ptime:20 >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:18 G729/8000 >> m=audio 31632 RTP/AVP 0 18 101 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849875 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+ >> 14792460547 at flowroute.com entering state [received][100] >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 >> (sofia/external/+14792460547 at flowroute.com) State NEW >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com [BREAK] >> 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel >> sofia/external/+14792460547 at flowroute.com[d276f012-5b65-11e2-beae-8564ff827a25] >> >> Robert >> >> >> ------------------------------ >> Date: Thu, 10 Jan 2013 09:47:18 -0800 >> From: msc at freeswitch.org >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Can you pastebin a console debug log w/ sip trace of a call failing to >> negotiate t.38 correctly? That might help us figure out where to look. >> -MC >> >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: >> >> Just installed FusionPBX using the script built for a Centos 6 install. >> Everything works great in regards to calls, but my main goal of this >> project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have >> tried many things in media bypass, routing to a fax extension, using an >> ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me >> in the right direction as to where I should begin? I am guessing it isn't >> supposed to work right out of the box. I am using Flowroute as my provider >> right now and the have confirmed that they utilize T38 in both directions. >> Thanks! >> >> >> Robert >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official FreeSWITCH >> Sites http://www.freeswitch.org http://wiki.freeswitch.org >> http://www.cluecon.com FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official FreeSWITCH >> Sites http://www.freeswitch.org http://wiki.freeswitch.org >> http://www.cluecon.com FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server Official FreeSWITCH >> Sites http://www.freeswitch.org http://wiki.freeswitch.org >> http://www.cluecon.com FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Profe >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/ac3c9494/attachment-0001.html From craig.robert at hotmail.com Fri Jan 18 17:21:40 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Fri, 18 Jan 2013 07:21:40 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: Message-ID: Ok, thanks. Robert On Jan 18, 2013, at 7:09 AM, "George Assaad" wrote: > Will check mine tonight > On Jan 18, 2013 8:41 AM, "Robert Craig" wrote: >> This was an incoming fax. I don't have much control over the other end. Why is that error appearing when trying to receive something? >> >> Robert >> >> >> >> On Jan 18, 2013, at 5:58 AM, "George Assaad" wrote: >> >>> Although the gui mentioned to attach a tiff, I finds it to not work. Please try pdf, you should be in business now. >>> Please let us know, >>> >>> On Jan 18, 2013 12:35 AM, "Robert Craig" wrote: >>>> OK, so I've made a little bit of progress. The distant end is a Windows Fax server hooked up to an analog line. I created a fax extension in FS, assigned an inbound route to the extension, and made sure the extension had an email address. I believe the fax receive wants to complete, but there is an interesting error in the log about TIF files. Anyone ever seen this before? >>>> >>>> v=0 >>>> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 >>>> s=FreeSWITCH >>>> c=IN IP4 72.201.112.131 >>>> t=0 0 >>>> m=image 27714 udptl t38 >>>> a=T38FaxVersion:0 >>>> a=T38MaxBitRate:9600 >>>> a=T38FaxFillBitRemoval >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> >>>> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. >>>> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === Negotiation Result ======================================================= >>>> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote station id: ATW-4792460547 >>>> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local station id: SpanDSP Fax Ident >>>> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer Rate: 9600 >>>> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM status off >>>> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote country: >>>> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote vendor: >>>> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote model: >>>> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 ============================================================================== >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (41) TIFF/F file cannot be opened. >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote station id: ATW-4792460547 >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 9600 >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM status off >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote country: >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote model: >>>> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== >>>> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. >>>> EXECUTE sofia/external/+14792460547 at flowroute.com hangup() >>>> >>>> Robert >>>> >>>> >>>> >>>> From: craig.robert at hotmail.com >>>> To: freeswitch-users at lists.freeswitch.org >>>> Date: Wed, 16 Jan 2013 10:40:31 -0700 >>>> Subject: Re: [Freeswitch-users] T38 Configuration >>>> >>>> OK, I am not extremely concerned with getting an ATA working right now. If I could just get an inbound extension-to-email working it would be great. I'll start from scratch and then let you know what happens. Just to make sure, I am going to create an extension, assign an email address to it, make sure there is an inbound route to that extension, and then test it. >>>> >>>> >>>> Robert >>>> >>>> >>>> Date: Wed, 16 Jan 2013 12:11:34 -0500 >>>> From: gassaad at emassembly.com >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] T38 Configuration >>>> >>>> The reason for outbound extension is: >>>> When you send a fax from ATA to the outside world, it will dial first to the outbound extension and then after successful receive it will dial out to the intended number. >>>> >>>> As for the error you have it may be related to your ATA. I am not sure please see why do you get the following message. >>>> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " >>>> >>>> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig wrote: >>>> OK, I've already tried creating one extension for inbound and set an email address. Of course, it failed and the errors are at the bottom of this email. What is the purpose of the outbound extension? >>>> >>>> Robert >>>> >>>> >>>> Date: Wed, 16 Jan 2013 09:57:51 -0500 >>>> From: gassaad at emassembly.com >>>> >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] T38 Configuration >>>> >>>> Sorry for the delay to get back to you. >>>> 1- App --> fax server >>>> 2- Create two extensions (one for inbound) another for (outbound) From within either extension you can send faxes "virtual fax" >>>> 3- point your create your outbound and inbound dialplans as normal but make it to point to those extensions. >>>> >>>> Good luck, >>>> >>>> George >>>> >>>> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig wrote: >>>> What did you do on the config side to get it to work? I've tried both an Ata and a fax to email extension and they both fail in both directions. >>>> >>>> Robert >>>> >>>> On Jan 12, 2013, at 2:04 PM, "George Assaad" wrote: >>>> >>>> I use Fusion successfully to send / receive fax. >>>> The way it works: >>>> ATA --->FaxServer. >>>> After successful receipt, Fax Server---> DID where you have dialled originally. >>>> >>>> Cheers, >>>> >>>> George >>>> >>>> >>>> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig wrote: >>>> Did anyone see the issue or what I may have missed in the config? As mentioned earlier, I thought it would after a fresh install. Any help is appreciated. >>>> >>>> Robert >>>> >>>> >>>> From: craig.robert at hotmail.com >>>> To: freeswitch-users at lists.freeswitch.org >>>> Date: Thu, 10 Jan 2013 13:43:09 -0700 >>>> >>>> Subject: Re: [Freeswitch-users] T38 Configuration >>>> >>>> Thanks for the quick response. Below is the capture from the log viewer. In this test, I am using a remote fax machine (line from Cox with physical machine) faxing into the "email server" where I created an extension for fax-to-email. Calling number is 1479-246-0547. >>>> >>>> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY going to sleep >>>> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:99 sofia/external/+14792460547 at flowroute.com Standard DESTROY >>>> 2013-01-10 13:39:53.162506 [DEBUG] mod_sofia.c:396 sofia/external/+14792460547 at flowroute.com SOFIA DESTROY >>>> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:569 (sofia/external/+14792460547 at flowroute.com) State DESTROY >>>> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:559 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_DESTROY >>>> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:556 (sofia/external/+14792460547 at flowroute.com) Callstate Change HANGUP -> DOWN >>>> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1510 Close Channel sofia/external/+14792460547 at flowroute.com [CS_DESTROY] >>>> 2013-01-10 13:39:53.162506 [NOTICE] switch_core_session.c:1506 Session 26 (sofia/external/+14792460547 at flowroute.com) Ended >>>> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1488 Session 26 (sofia/external/+14792460547 at flowroute.com) Locked, Waiting on external entities >>>> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:440 (sofia/external/+14792460547 at flowroute.com) State Change CS_REPORTING -> CS_DESTROY >>>> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING going to sleep >>>> 2013-01-10 13:39:53.162506 [DEBUG] switch_core_state_machine.c:92 sofia/external/+14792460547 at flowroute.com Standard REPORTING, cause: NORMAL_CLEARING >>>> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:749 (sofia/external/+14792460547 at flowroute.com) State REPORTING >>>> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_REPORTING >>>> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:446 (sofia/external/+14792460547 at flowroute.com) State Change CS_HANGUP -> CS_REPORTING >>>> >>>> 2013-01-10 13:39:53.066135 [DEBUG] switch_core_state_machine.c:599 Hangup Command with no Session system(/usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=+14792460547-2013-01-10-13-39-26 messages='result: Disconnected after permitted retries sender:ATW-4792460547 pages:0' domain=192.168.4.15 caller_id_name='' caller_id_number=): >>>> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP going to sleep >>>> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:48 sofia/external/+14792460547 at flowroute.com Standard HANGUP, cause: NORMAL_CLEARING >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:552 Sending BYE to sofia/external/+14792460547 at flowroute.com >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_sofia.c:503 Channel sofia/external/+14792460547 at flowroute.com hanging up, cause: NORMAL_CLEARING >>>> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:667 (sofia/external/+14792460547 at flowroute.com) State HANGUP >>>> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_HANGUP >>>> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep >>>> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:2687 sofia/external/+14792460547 at flowroute.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >>>> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:3017 Send signal sofia/external/+14792460547 at flowroute.com [KILL] >>>> 2013-01-10 13:39:52.216151 [NOTICE] mod_dptools.c:1150 Hangup sofia/external/+14792460547 at flowroute.com [CS_EXECUTE] [NORMAL_CLEARING] >>>> 2013-01-10 13:39:52.216151 [DEBUG] switch_channel.c:2994 (sofia/external/+14792460547 at flowroute.com) Callstate Change ACTIVE -> HANGUP >>>> EXECUTE sofia/external/+14792460547 at flowroute.com hangup() >>>> 2013-01-10 13:39:52.216151 [DEBUG] switch_core_codec.c:244 sofia/external/+14792460547 at flowroute.com Restore previous codec PCMU:0. >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:518 ============================================================================== >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:516 remote model: >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:515 remote vendor: >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:514 remote country: >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:513 ECM status off >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:511 Transfer Rate: 14400 >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:510 Image resolution: 0x0 >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:509 Total fax pages: 0 >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:507 Pages transferred: 0 >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:506 Local station id: SpanDSP Fax Ident >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:505 Remote station id: ATW-4792460547 >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (48) Disconnected after permitted retries. >>>> 2013-01-10 13:39:52.216151 [DEBUG] mod_spandsp_fax.c:487 ============================================================================== >>>> 2013-01-10 13:39:38.766190 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. >>>> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxFillBitRemoval >>>> a=T38MaxBitRate:9600 >>>> a=T38FaxVersion:0 >>>> m=image 27588 udptl t38 >>>> t=0 0 >>>> c=IN IP4 72.201.112.131 >>>> s=FreeSWITCH >>>> o=FreeSWITCH 1357822778 1357822782 IN IP4 72.201.112.131 >>>> v=0 >>>> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: >>>> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >>>> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >>>> 2013-01-10 13:39:29.576127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxFillBitRemoval >>>> a=T38MaxBitRate:9600 >>>> a=T38FaxVersion:0 >>>> m=image 27588 udptl t38 >>>> t=0 0 >>>> c=IN IP4 72.201.112.131 >>>> s=FreeSWITCH >>>> o=FreeSWITCH 1357822778 1357822781 IN IP4 72.201.112.131 >>>> v=0 >>>> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: >>>> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >>>> 2013-01-10 13:39:29.576127 [DEBUG] mod_sofia.c:1487 Remote address:port [66.42.121.5:31632] has not changed. >>>> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 >>>> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 >>>> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 >>>> 2013-01-10 13:39:29.576127 [ERR] sofia_glue.c:3549 Invalid Jitterbuffer spec [0] must be between 20 and 10000 >>>> 2013-01-10 13:39:29.576127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms >>>> 2013-01-10 13:39:29.576127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 >>>> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] >>>> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:29.566140 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> >>>> a=T38FaxRateManagement:transferredTCF >>>> m=image 31632 udptl t38 >>>> t=0 0 >>>> c=IN IP4 66.42.121.5 >>>> s=- >>>> o=- 275426305 1357849876 IN IP4 66.42.121.5 >>>> v=0 >>>> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 Remote SDP: >>>> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completing][200] >>>> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 sofia/external/+14792460547 at flowroute.com Update Callee ID to "+14792460547" <+14792460547> >>>> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:29.556130 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [calling][0] >>>> 2013-01-10 13:39:29.256196 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxFillBitRemoval >>>> a=T38MaxBitRate:9600 >>>> a=T38FaxVersion:0 >>>> m=image 27588 udptl t38 >>>> t=0 0 >>>> c=IN IP4 72.201.112.131 >>>> s=FreeSWITCH >>>> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >>>> v=0 >>>> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:2647 Local SDP: >>>> >>>> a=T38FaxUdpEC:t38UDPRedundancy >>>> a=T38FaxMaxDatagram:400 >>>> a=T38FaxMaxBuffer:2000 >>>> a=T38FaxRateManagement:transferredTCF >>>> a=T38FaxFillBitRemoval >>>> a=T38MaxBitRate:9600 >>>> a=T38FaxVersion:0 >>>> m=image 27588 udptl t38 >>>> t=0 0 >>>> c=IN IP4 72.201.112.131 >>>> s=FreeSWITCH >>>> o=FreeSWITCH 1357822778 1357822780 IN IP4 72.201.112.131 >>>> v=0 >>>> 2013-01-10 13:39:29.256196 [DEBUG] sofia_glue.c:173 sofia/external/+14792460547 at flowroute.com image media sdp: >>>> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1379 Raw write codec activation Success L16 >>>> 2013-01-10 13:39:27.996269 [DEBUG] switch_core_codec.c:219 sofia/external/+14792460547 at flowroute.com Push codec L16:70 >>>> 2013-01-10 13:39:27.996269 [DEBUG] mod_spandsp_fax.c:1363 Raw read codec activation Success L16 20000 >>>> EXECUTE sofia/external/+14792460547 at flowroute.com rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) >>>> 2013-01-10 13:39:27.996269 [DEBUG] switch_ivr_play_say.c:1682 done playing file silence_stream://2000 >>>> 2013-01-10 13:39:26.256197 [DEBUG] switch_rtp.c:3624 Correct ip/port confirmed. >>>> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [ready][200] >>>> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:26.096329 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:26.016177 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms >>>> EXECUTE sofia/external/+14792460547 at flowroute.com playback(silence_stream://2000) >>>> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [last_fax]=[+14792460547-2013-01-10-13-39-26] >>>> EXECUTE sofia/external/+14792460547 at flowroute.com set(last_fax=+14792460547-2013-01-10-13-39-26) >>>> 2013-01-10 13:39:26.016177 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38_request]=[true] >>>> EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38_request=true) >>>> 2013-01-10 13:39:26.006127 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [fax_enable_t38]=[true] >>>> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [completed][200] >>>> EXECUTE sofia/external/+14792460547 at flowroute.com set(fax_enable_t38=true) >>>> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:26.006127 [NOTICE] mod_dptools.c:1176 Channel [sofia/external/+14792460547 at flowroute.com] has been answered >>>> 2013-01-10 13:39:26.006127 [DEBUG] switch_channel.c:3395 (sofia/external/+14792460547 at flowroute.com) Callstate Change RINGING -> ACTIVE >>>> 2013-01-10 13:39:26.006127 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> >>>> a=sendrecv >>>> a=ptime:20 >>>> a=silenceSupp:off - - - - >>>> a=fmtp:101 0-16 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=rtpmap:0 PCMU/8000 >>>> m=audio 27588 RTP/AVP 0 101 >>>> t=0 0 >>>> c=IN IP4 72.201.112.131 >>>> s=FreeSWITCH >>>> o=FreeSWITCH 1357822778 1357822779 IN IP4 72.201.112.131 >>>> v=0 >>>> 2013-01-10 13:39:26.006127 [DEBUG] mod_sofia.c:856 Local SDP sofia/external/+14792460547 at flowroute.com: >>>> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3645 sofia/external/+14792460547 at flowroute.com Set rtp dtmf delay to 40 >>>> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3618 Set 2833 dtmf receive payload to 101 >>>> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3612 Set 2833 dtmf send payload to 101 >>>> 2013-01-10 13:39:26.006127 [DEBUG] switch_rtp.c:1928 Starting timer [soft] 160 bytes per 20ms >>>> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:3348 AUDIO RTP [sofia/external/+14792460547 at flowroute.com] 192.168.4.15 port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 >>>> 2013-01-10 13:39:26.006127 [DEBUG] sofia_glue.c:1046 STUN Success [72.201.112.131]:[27588] >>>> EXECUTE sofia/external/+14792460547 at flowroute.com answer() >>>> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [api_hangup_hook]=[system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=${last_fax} messages='result: ${fax_result_text} sender:${fax_remote_station_id} pages:${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='${caller_id_name}' caller_id_number=${caller_id_number}] >>>> EXECUTE sofia/external/+14792460547 at flowroute.com set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\${last_fax} messages='result: \${fax_result_text} sender:\${fax_remote_station_id} pages:\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\${caller_id_name}' caller_id_number=\${caller_id_number}) >>>> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] >>>> EXECUTE sofia/external/+14792460547 at flowroute.com set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) >>>> EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) >>>> EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-last_dial/+14792460547/3150) >>>> EXECUTE sofia/external/+14792460547 at flowroute.com hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) >>>> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [call_direction]=[local] >>>> EXECUTE sofia/external/+14792460547 at flowroute.com set(call_direction=local) >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE >>>> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action hangup() >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action playback(silence_stream://2000) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38_request=true) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(fax_enable_t38=true) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action answer() >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(api_hangup_hook=system /usr/bin/php /var/www/html/fusionpbx/secure/fax_to_email.php email=robert.l.craig at craignetwork.com extension=3150 name=\\\${last_fax} messages='result: \\\${fax_result_text} sender:\\\${fax_remote_station_id} pages:\\\${fax_document_total_pages}' domain=192.168.4.15 caller_id_name='\\\${caller_id_name}' caller_id_number=\\\${caller_id_number}) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test Fax] destination_number(3150) =~ /^3150$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Test Fax] continue=false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain_user] destination_number(3150) =~ /^\*97$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain_user] continue=false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [vmain] destination_number(3150) =~ /^vmain$|^\*4000$|^\*98$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->vmain] continue=false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [send_to_voicemail] destination_number(3150) =~ /^\*99(\d{2,7})$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->send_to_voicemail] continue=false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [extension-intercom] destination_number(3150) =~ /^\*8(\d{2,7})$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->extension-intercom] continue=false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [intercept-ext] destination_number(3150) =~ /^\*\*(\d+)$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->intercept-ext] continue=false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_return] destination_number(3150) =~ /^\*69$|^lcr$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_return] continue=false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_privacy] destination_number(3150) =~ /^\*67(\d+)$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call_privacy] continue=false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/global/${uuid}) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [global] >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->global] continue=true >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [redial] destination_number(3150) =~ /^(redial|\*870)$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->redial] continue=false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [group-intercept] destination_number(3150) =~ /^\*8$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->group-intercept] continue=false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com ANTI-Action set(call_direction=local) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call-direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->call-direction] continue=true >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [Flowroute.11d] destination_number(3150) =~ /^\+?(\d{11})$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [default->Flowroute.11d] continue=false >>>> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->3150 in context default >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING >>>> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE going to sleep >>>> 2013-01-10 13:39:25.816280 [NOTICE] switch_ivr.c:1781 Transfer sofia/external/+14792460547 at flowroute.com to XML[3150 at default] >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:830 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_ivr.c:1774 (sofia/external/+14792460547 at flowroute.com) State Change CS_EXECUTE -> CS_ROUTING >>>> EXECUTE sofia/external/+14792460547 at flowroute.com transfer(3150 XML default) >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] >>>> EXECUTE sofia/external/+14792460547 at flowroute.com export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) >>>> 2013-01-10 13:39:25.816280 [DEBUG] mod_dptools.c:1344 sofia/external/+14792460547 at flowroute.com SET [outside_call]=[true] >>>> EXECUTE sofia/external/+14792460547 at flowroute.com set(outside_call=true) >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:209 sofia/external/+14792460547 at flowroute.com Standard EXECUTE >>>> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:242 sofia/external/+14792460547 at flowroute.com SOFIA EXECUTE >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:477 (sofia/external/+14792460547 at flowroute.com) State EXECUTE >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_EXECUTE >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING going to sleep >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:167 (sofia/external/+14792460547 at flowroute.com) State Change CS_ROUTING -> CS_EXECUTE >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action transfer(3150 XML default) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] destination_number(15203039810) =~ /^15203039810$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [Test_Number] context(public) =~ /public/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->Test_Number] continue=false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->call_debug] continue=true >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Action set(outside_call=true) >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Absolute Condition [outside_call] >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->outside_call] continue=true >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >>>> Dialplan: sofia/external/+14792460547 at flowroute.com parsing [public->unloop] continue=false >>>> 2013-01-10 13:39:25.816280 [INFO] mod_dialplan_xml.c:498 Processing +14792460547 <+14792460547>->15203039810 in context public >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:117 sofia/external/+14792460547 at flowroute.com Standard ROUTING >>>> 2013-01-10 13:39:25.816280 [DEBUG] mod_sofia.c:149 sofia/external/+14792460547 at flowroute.com SOFIA ROUTING >>>> 2013-01-10 13:39:25.816280 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+14792460547 at flowroute.com) State ROUTING >>>> 2013-01-10 13:39:25.806120 [DEBUG] switch_channel.c:2003 (sofia/external/+14792460547 at flowroute.com) Callstate Change DOWN -> RINGING >>>> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_ROUTING >>>> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT going to sleep >>>> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:126 (sofia/external/+14792460547 at flowroute.com) State Change CS_INIT -> CS_ROUTING >>>> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 sofia/external/+14792460547 at flowroute.com SOFIA INIT >>>> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+14792460547 at flowroute.com) State INIT >>>> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_INIT >>>> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_session.c:1283 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 (sofia/external/+14792460547 at flowroute.com) State Change CS_NEW -> CS_INIT >>>> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5263 Set 2833 dtmf send/recv payload to 101 >>>> 2013-01-10 13:39:25.806120 [DEBUG] switch_core_codec.c:111 sofia/external/+14792460547 at flowroute.com Original read codec set to PCMU:0 >>>> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:3093 Set Codec sofia/external/+14792460547 at flowroute.com PCMU/8000 20 ms 160 samples 64000 bits >>>> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >>>> 2013-01-10 13:39:25.806120 [DEBUG] sofia_glue.c:5134 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >>>> >>>> a=ptime:20 >>>> a=fmtp:101 0-15 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:18 annexb=no >>>> a=rtpmap:18 G729/8000 >>>> m=audio 31632 RTP/AVP 0 18 101 >>>> t=0 0 >>>> c=IN IP4 66.42.121.5 >>>> s=- >>>> o=- 275426305 1357849875 IN IP4 66.42.121.5 >>>> v=0 >>>> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 Remote SDP: >>>> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 Channel sofia/external/+14792460547 at flowroute.com entering state [received][100] >>>> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:433 (sofia/external/+14792460547 at flowroute.com) State NEW >>>> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+14792460547 at flowroute.com) Running State Change CS_NEW >>>> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:25.786370 [DEBUG] switch_core_session.c:975 Send signal sofia/external/+14792460547 at flowroute.com [BREAK] >>>> 2013-01-10 13:39:25.786370 [NOTICE] switch_channel.c:968 New Channel sofia/external/+14792460547 at flowroute.com [d276f012-5b65-11e2-beae-8564ff827a25] >>>> >>>> Robert >>>> >>>> >>>> Date: Thu, 10 Jan 2013 09:47:18 -0800 >>>> From: msc at freeswitch.org >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] T38 Configuration >>>> >>>> Can you pastebin a console debug log w/ sip trace of a call failing to negotiate t.38 correctly? That might help us figure out where to look. >>>> -MC >>>> >>>> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig wrote: >>>> Just installed FusionPBX using the script built for a Centos 6 install. Everything works great in regards to calls, but my main goal of this project was to utilize the fax capabilities of Freeswitch/FusionPBX. I have tried many things in media bypass, routing to a fax extension, using an ATA, etc., but I can't get T38 to negotiate correctly. Can someone point me in the right direction as to where I should begin? I am guessing it isn't supposed to work right out of the box. I am using Flowroute as my provider right now and the have confirmed that they utilize T38 in both directions. Thanks! >>>> >>>> >>>> Robert >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ Profe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> ... > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/a7e2be10/attachment-0001.html -------------- next part -------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From cal.leeming at simplicitymedialtd.co.uk Fri Jan 18 17:31:06 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 18 Jan 2013 14:31:06 +0000 Subject: [Freeswitch-users] NAT traversal - the final say..! In-Reply-To: <1357229550070-7586032.post@n2.nabble.com> References: <1357229550070-7586032.post@n2.nabble.com> Message-ID: Hi all, Thanks for everyones feedback on this, there's a lot of really useful info and this will make life much easier when writing the docs. I'm hoping to make a start on this within the next month (mod_xml_curl and mod_event_socket are first in my list). Of course, others are more than welcome to contribute as well :) Cal On Thu, Jan 3, 2013 at 4:12 PM, peely wrote: > IMO as a provider, FS does an excellent job at detecting NAT on its own and > handling it. We've rarely had any issue with our end sending INVITEs to the > wrong place or one / no way speech. connectile dysfunction does nothing > more. The only switch we enable is one to enable the SIP options ping and > to > use sip-force-expires to reduce the time the endpoint re-registers for > mobile devices. > > As with others on this thread, the ALG in most routers is the culprit, I've > rarely experienced one which does anything beneficial, most screw up the > headers. > > We also recommend that customers change the SIP port on their handsets to > anything other than 5060, older BT routers especially seem to stop > forwarding responses after a few minutes, again as they have a very basic > ALG. Using port forwarding in the router is the only sure fire way of > getting reliable SIP through to a handset, Draytek routers seem to be > pretty > flaky until you do this or use TCP / TLS. > > If the router stops forwarding packets because of a poor ALG implementation > or other firewall rules, nothing FS can do will overcome this. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/NAT-traversal-the-final-say-tp7585562p7586032.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/dc748a8b/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri Jan 18 17:33:12 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 18 Jan 2013 14:33:12 +0000 Subject: [Freeswitch-users] mod_xml_curl - request for comments Message-ID: Hello, I've been re-reading my changes to mod_xml_curl documentation, and I'm still not entirely happy with it. I can't tell if it genuinely needs a different approach, or if I'm just being fussy. Therefore, constructive criticism is very much welcome - even if it means the whole damn thing has to be re-done again. http://wiki.freeswitch.org/wiki/Mod_xml_curl Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/19da4713/attachment.html From admin at smallunix.net Fri Jan 18 17:35:41 2013 From: admin at smallunix.net (Andrea Mazzeo) Date: Fri, 18 Jan 2013 15:35:41 +0100 Subject: [Freeswitch-users] SONUS - dtg inside INVITE In-Reply-To: References: <7443520999689776200@unknownmsgid> <5D31EF90-20B7-48BA-B7BC-65F30B5383DD@jerris.com> Message-ID: Dear all, I found the solution. sip_invite_contact_params=dtg=ATTMDT7807IPO I hope this can help other people. Best regards, Andrea 2013/1/18 Andrea Mazzeo : > Hi Michael, > > In the contact field. > > 2013/1/18 Michael Jerris : >> What you posted is NOT a valid invite header. WHERE in the invite header do you need it? >> >> On Jan 17, 2013, at 6:59 PM, Andrea Mazzeo wrote: >> >>> Hi Kristian, >>> >>> I need dtg in the INVITE header. >>> >>> Best regards, >>> Andrea Mazzeo >>> >>> Il giorno 17/gen/2013, alle ore 22:01, Kristian Kielhofner >>> ha scritto: >>> >>>> This is almost certainly not the syntax you want to use. >>>> >>>> Do that expect "dtg" to be a user or domain parameter? >>>> >>>> On Thu, Jan 17, 2013 at 1:46 PM, Andrea Mazzeo wrote: >>>>> Hi all, >>>>> >>>>> I need to configure my FS to send call to a Sonus GSX via SIP. >>>>> Is there a way to send SIP parameter "dtg" (destination trunk group) >>>>> inside the INVITE? >>>>> >>>>> Example: >>>>> >>>>> Request-Line: INVITE sip: 39340xxxxxx at 38.xxx.xxx.xxx SIP/2.0 dtg=ATTMDT7807IPO >>>>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org From kris at kriskinc.com Fri Jan 18 17:36:58 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 18 Jan 2013 09:36:58 -0500 Subject: [Freeswitch-users] SONUS - dtg inside INVITE In-Reply-To: References: <7443520999689776200@unknownmsgid> <5D31EF90-20B7-48BA-B7BC-65F30B5383DD@jerris.com> Message-ID: Are you sure about that? 99% of the time this stuff goes in the destination URI (either the host or user portion), which leads back to my original question... On Fri, Jan 18, 2013 at 3:01 AM, Andrea Mazzeo wrote: > Hi Michael, > > In the contact field. > > 2013/1/18 Michael Jerris : >> What you posted is NOT a valid invite header. WHERE in the invite header do you need it? >> >> On Jan 17, 2013, at 6:59 PM, Andrea Mazzeo wrote: >> >>> Hi Kristian, >>> >>> I need dtg in the INVITE header. >>> >>> Best regards, >>> Andrea Mazzeo >>> >>> Il giorno 17/gen/2013, alle ore 22:01, Kristian Kielhofner >>> ha scritto: >>> >>>> This is almost certainly not the syntax you want to use. >>>> >>>> Do that expect "dtg" to be a user or domain parameter? >>>> >>>> On Thu, Jan 17, 2013 at 1:46 PM, Andrea Mazzeo wrote: >>>>> Hi all, >>>>> >>>>> I need to configure my FS to send call to a Sonus GSX via SIP. >>>>> Is there a way to send SIP parameter "dtg" (destination trunk group) >>>>> inside the INVITE? >>>>> >>>>> Example: >>>>> >>>>> Request-Line: INVITE sip: 39340xxxxxx at 38.xxx.xxx.xxx SIP/2.0 dtg=ATTMDT7807IPO >>>>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From Tim.Meade at Millicorp.com Fri Jan 18 17:50:13 2013 From: Tim.Meade at Millicorp.com (Tim Meade) Date: Fri, 18 Jan 2013 14:50:13 +0000 Subject: [Freeswitch-users] mod_xml_curl - request for comments In-Reply-To: References: Message-ID: <804D48104511D4468F0D60DF9D3100350B0A1102@MAIL.millicorp.com> Hi Cal.... I think it looks MUCH better. Great job, Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cal Leeming [Simplicity Media Ltd] Sent: Friday, January 18, 2013 9:33 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] mod_xml_curl - request for comments Hello, I've been re-reading my changes to mod_xml_curl documentation, and I'm still not entirely happy with it. I can't tell if it genuinely needs a different approach, or if I'm just being fussy. Therefore, constructive criticism is very much welcome - even if it means the whole damn thing has to be re-done again. http://wiki.freeswitch.org/wiki/Mod_xml_curl Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/5eed49d9/attachment-0001.html From admin at smallunix.net Fri Jan 18 17:46:40 2013 From: admin at smallunix.net (Andrea Mazzeo) Date: Fri, 18 Jan 2013 15:46:40 +0100 Subject: [Freeswitch-users] SONUS - dtg inside INVITE In-Reply-To: References: <7443520999689776200@unknownmsgid> <5D31EF90-20B7-48BA-B7BC-65F30B5383DD@jerris.com> Message-ID: Hi Kristian, I did a trace, and it's in the Contact URI. 2013/1/18 Kristian Kielhofner : > Are you sure about that? > > 99% of the time this stuff goes in the destination URI (either the > host or user portion), which leads back to my original question... > > On Fri, Jan 18, 2013 at 3:01 AM, Andrea Mazzeo wrote: >> Hi Michael, >> >> In the contact field. >> >> 2013/1/18 Michael Jerris : >>> What you posted is NOT a valid invite header. WHERE in the invite header do you need it? >>> >>> On Jan 17, 2013, at 6:59 PM, Andrea Mazzeo wrote: >>> >>>> Hi Kristian, >>>> >>>> I need dtg in the INVITE header. >>>> >>>> Best regards, >>>> Andrea Mazzeo >>>> >>>> Il giorno 17/gen/2013, alle ore 22:01, Kristian Kielhofner >>>> ha scritto: >>>> >>>>> This is almost certainly not the syntax you want to use. >>>>> >>>>> Do that expect "dtg" to be a user or domain parameter? >>>>> >>>>> On Thu, Jan 17, 2013 at 1:46 PM, Andrea Mazzeo wrote: >>>>>> Hi all, >>>>>> >>>>>> I need to configure my FS to send call to a Sonus GSX via SIP. >>>>>> Is there a way to send SIP parameter "dtg" (destination trunk group) >>>>>> inside the INVITE? >>>>>> >>>>>> Example: >>>>>> >>>>>> Request-Line: INVITE sip: 39340xxxxxx at 38.xxx.xxx.xxx SIP/2.0 dtg=ATTMDT7807IPO >>>>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vbvbrj at gmail.com Fri Jan 18 17:58:49 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 18 Jan 2013 16:58:49 +0200 Subject: [Freeswitch-users] Nested Condition Message-ID: <50F96329.9050007@gmail.com> Hi. There is scenario: Calling "pref1"-"ext" must redirect (transfer) to "ext" if "ext" exists in xml directory, ie user_exists(id ext domain). If user does not exists, dialplan processing must continue. In the same extension also calling "pref2"-"ext" must be processed same way. But it is all to $1 scope. I try: I know this is incorrect. But how to do this? -- Mimiko desu. From support at ecn.net.au Fri Jan 18 18:20:28 2013 From: support at ecn.net.au (support at ecn.net.au) Date: Sat, 19 Jan 2013 01:20:28 +1000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <20130118102614.GA27116@eagle.cupis.co.uk> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> , <20130118102614.GA27116@eagle.cupis.co.uk> Message-ID: <8A80C99638A5E24E83696EAE33A9B6B968B5302824@twbexmbx001.ecn.hosting> Yes, we read that and tried to set that when attempting multiple configurations (attempting to pass inband dtmf). Currently we can't get Freeswitch in an SBC role to pass the inband in the media stream at all (see earlier email) - also having odd issues with start_dtmf when accessing Asterisk IVR's (again very odd). the only difference we can see (between FS and our legacy openser) is the Invite pushed on the bleg on Openser included the fmtp:18 annexb=no directive (as per the telco's SDP to us); however on FS it doesn't forward these (and we can't seem to force FS to add these attributes to the SDP on the bleg). Again the start_dtmf issues exists only in 1.2 and 1.3 FS , in the 1.0.6 the issue is not present (is this a bug possibly?) Any ideas? Kind Regards, Mark ________________________________________ From: Paul Cupis [paul at cupis.co.uk] Sent: Friday, 18 January 2013 8:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SBC In-band DTMF On Fri, Jan 18, 2013 at 01:57:36AM +0000, Steven Ayre wrote: > > > What're you hoping to do here? This sets the payload type value. 0 is a > valid pt. This won't disable it, it'll instead use the value 0. That > happens to be the one for G711, so if that codec is enabled you're just > going to get a conflict. If the rfc2833-pt is set to a value <96 it is outside the allowed range for dynamic payload negotiation and FS will not offer RFC2833 support in the SDP. ref: sofia_glue.c Regards, From steveayre at gmail.com Fri Jan 18 19:05:26 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Jan 2013 16:05:26 +0000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <8A80C99638A5E24E83696EAE33A9B6B968B5302818@twbexmbx001.ecn.hosting> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B5302818@twbexmbx001.ecn.hosting> Message-ID: There is no way in SIP to tell someone that you are inband tones. Rather you tell them you're using out-of-band RFC2833 or INFO. On 18 January 2013 05:19, support at ecn.net.au wrote: > (perhaps not telling the PBX that > we are using inband). > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/aa2c772c/attachment.html From steveayre at gmail.com Fri Jan 18 19:06:05 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Jan 2013 16:06:05 +0000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <20130118102614.GA27116@eagle.cupis.co.uk> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> <20130118102614.GA27116@eagle.cupis.co.uk> Message-ID: Static payload values (<96) can be reassigned dynamically with a=rtpmap lines, but could perhaps conflict with loaded modules already using those values. On 18 January 2013 10:26, Paul Cupis wrote: > On Fri, Jan 18, 2013 at 01:57:36AM +0000, Steven Ayre wrote: > > > > > > What're you hoping to do here? This sets the payload type value. 0 is > a > > valid pt. This won't disable it, it'll instead use the value 0. That > > happens to be the one for G711, so if that codec is enabled you're > just > > going to get a conflict. > > If the rfc2833-pt is set to a value <96 it is outside the allowed range > for dynamic payload negotiation and FS will not offer RFC2833 support > in the SDP. > > ref: sofia_glue.c > > Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/aaebd647/attachment.html From steveayre at gmail.com Fri Jan 18 19:06:59 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Jan 2013 16:06:59 +0000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <8A80C99638A5E24E83696EAE33A9B6B968B5302824@twbexmbx001.ecn.hosting> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> <20130118102614.GA27116@eagle.cupis.co.uk> <8A80C99638A5E24E83696EAE33A9B6B968B5302824@twbexmbx001.ecn.hosting> Message-ID: I would capture packet traces of SIP from FS->PBX under 1.0.6 and 1.2 and compare those here. On 18 January 2013 15:20, support at ecn.net.au wrote: > Again the start_dtmf issues exists only in 1.2 and 1.3 FS , in the 1.0.6 > the issue is not present (is this a bug possibly?) > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/1190224d/attachment.html From kris at kriskinc.com Fri Jan 18 19:08:54 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 18 Jan 2013 11:08:54 -0500 Subject: [Freeswitch-users] SONUS - dtg inside INVITE In-Reply-To: References: <7443520999689776200@unknownmsgid> <5D31EF90-20B7-48BA-B7BC-65F30B5383DD@jerris.com> Message-ID: Very interesting... If you don't mind saying, what provider is this? On Fri, Jan 18, 2013 at 9:46 AM, Andrea Mazzeo wrote: > Hi Kristian, > > I did a trace, and it's in the Contact URI. > > > 2013/1/18 Kristian Kielhofner : >> Are you sure about that? >> >> 99% of the time this stuff goes in the destination URI (either the >> host or user portion), which leads back to my original question... >> >> On Fri, Jan 18, 2013 at 3:01 AM, Andrea Mazzeo wrote: >>> Hi Michael, >>> >>> In the contact field. >>> >>> 2013/1/18 Michael Jerris : >>>> What you posted is NOT a valid invite header. WHERE in the invite header do you need it? >>>> >>>> On Jan 17, 2013, at 6:59 PM, Andrea Mazzeo wrote: >>>> >>>>> Hi Kristian, >>>>> >>>>> I need dtg in the INVITE header. >>>>> >>>>> Best regards, >>>>> Andrea Mazzeo >>>>> >>>>> Il giorno 17/gen/2013, alle ore 22:01, Kristian Kielhofner >>>>> ha scritto: >>>>> >>>>>> This is almost certainly not the syntax you want to use. >>>>>> >>>>>> Do that expect "dtg" to be a user or domain parameter? >>>>>> >>>>>> On Thu, Jan 17, 2013 at 1:46 PM, Andrea Mazzeo wrote: >>>>>>> Hi all, >>>>>>> >>>>>>> I need to configure my FS to send call to a Sonus GSX via SIP. >>>>>>> Is there a way to send SIP parameter "dtg" (destination trunk group) >>>>>>> inside the INVITE? >>>>>>> >>>>>>> Example: >>>>>>> >>>>>>> Request-Line: INVITE sip: 39340xxxxxx at 38.xxx.xxx.xxx SIP/2.0 dtg=ATTMDT7807IPO >>>>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Kristian Kielhofner >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From steveayre at gmail.com Fri Jan 18 19:12:57 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Jan 2013 16:12:57 +0000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <8A80C99638A5E24E83696EAE33A9B6B968B5302824@twbexmbx001.ecn.hosting> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> <20130118102614.GA27116@eagle.cupis.co.uk> <8A80C99638A5E24E83696EAE33A9B6B968B5302824@twbexmbx001.ecn.hosting> Message-ID: Did you try removing the rfc2833-pt param and setting on the outgoing sofia profile to the PBX? Inband tones will always be passed over, FS would actively need to detect the tone and remove it from the media to avoid doing so. So the PBX is not looking inband for it. Since there's no way in SIP to notify that inband tones are in use the PBX is probably checking for out-of-band DTMF and not checking inband at all if out-of-band has been negotiated - so what you probably need to is disable out of band media entirely. As far as comparing FS behaviour to OpenSER goes I would avoid it. They're very different products. Understand the difference between a SIP proxy (OpenSER) and a B2BUA (FreeSWITCH). OpenSER only forwards SIP messages with media terminating on the endpoints. FS negotiates 2 separate calls including media and terminates both calls on FS and passes media between them. As such OpenSER doesn't do any media negotiation and FS does which will make their behaviour rather different. -Steve On 18 January 2013 15:20, support at ecn.net.au wrote: > Yes, we read that and tried to set that when attempting multiple > configurations (attempting to pass inband dtmf). > > Currently we can't get Freeswitch in an SBC role to pass the inband in the > media stream at all (see earlier email) - also having odd issues with > start_dtmf when accessing Asterisk IVR's (again very odd). the only > difference we can see (between FS and our legacy openser) is the Invite > pushed on the bleg on Openser included the fmtp:18 annexb=no directive (as > per the telco's SDP to us); however on FS it doesn't forward these (and we > can't seem to force FS to add these attributes to the SDP on the bleg). > > Again the start_dtmf issues exists only in 1.2 and 1.3 FS , in the 1.0.6 > the issue is not present (is this a bug possibly?) > > Any ideas? > > Kind Regards, > > Mark > > ________________________________________ > From: Paul Cupis [paul at cupis.co.uk] > Sent: Friday, 18 January 2013 8:26 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SBC In-band DTMF > > On Fri, Jan 18, 2013 at 01:57:36AM +0000, Steven Ayre wrote: > > > > > > What're you hoping to do here? This sets the payload type value. 0 is > a > > valid pt. This won't disable it, it'll instead use the value 0. That > > happens to be the one for G711, so if that codec is enabled you're > just > > going to get a conflict. > > If the rfc2833-pt is set to a value <96 it is outside the allowed range > for dynamic payload negotiation and FS will not offer RFC2833 support > in the SDP. > > ref: sofia_glue.c > > Regards, > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/35b33ade/attachment.html From steveu at coppice.org Fri Jan 18 19:18:45 2013 From: steveu at coppice.org (Steve Underwood) Date: Sat, 19 Jan 2013 00:18:45 +0800 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: Message-ID: <50F975E5.2040206@coppice.org> If the TIFF file cannot be opened on receive you probably have a file permissions issue (e.g. with the directory in which the file is to be created). If the TIFF file cannot be opened on transmit there are various possibilities about the suitability of the file's contents. Steve On 01/18/2013 09:37 PM, Robert Craig wrote: > This was an incoming fax. I don't have much control over the other > end. Why is that error appearing when trying to receive something? > > Robert > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > wrote: > >> Although the gui mentioned to attach a tiff, I finds it to not work. >> Please try pdf, you should be in business now. >> Please let us know, >> >> On Jan 18, 2013 12:35 AM, "Robert Craig" > > wrote: >> >> OK, so I've made a little bit of progress. The distant end is a >> Windows Fax server hooked up to an analog line. I created a fax >> extension in FS, assigned an inbound route to the extension, and >> made sure the extension had an email address. I believe the fax >> receive wants to complete, but there is an interesting error in >> the log about TIF files. Anyone ever seen this before? >> >> v=0 >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 >> s=FreeSWITCH >> c=IN IP4 72.201.112.131 >> t=0 0 >> m=image 27714 udptl t38 >> a=T38FaxVersion:0 >> a=T38MaxBitRate:9600 >> a=T38FaxFillBitRemoval >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxMaxBuffer:2000 >> a=T38FaxMaxDatagram:400 >> a=T38FaxUdpEC:t38UDPRedundancy >> >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 >> @flowroute.com [BREAK] >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send >> signal sofia/external/+14792460547 >> @flowroute.com [BREAK] >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct >> ip/port confirmed. >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === >> Negotiation Result >> ======================================================= >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote >> station id: ATW-4792460547 >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local >> station id: SpanDSP Fax Ident >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer >> Rate: 9600 >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM >> status off >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote >> country: >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote >> vendor: >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote >> model: >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 >> ============================================================================== >> >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 >> ============================================================================== >> >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax >> processing not successful - result (41) TIFF/F file cannot be >> opened.* >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote >> station id: ATW-4792460547 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local >> station id: SpanDSP Fax Ident >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages >> transferred: 0 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total >> fax pages: 0 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image >> resolution: 0x0 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer >> Rate: 9600 >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM >> status off >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote >> country: >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote >> vendor: >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote >> model: >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 >> ============================================================================== >> >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 >> sofia/external/+14792460547 @flowroute.com >> Restore previous codec PCMU:0. >> EXECUTE sofia/external/+14792460547 >> @flowroute.com hangup() >> >> Robert >> >> >> >> ------------------------------------------------------------------------ >> From: craig.robert at hotmail.com >> To: freeswitch-users at lists.freeswitch.org >> >> Date: Wed, 16 Jan 2013 10:40:31 -0700 >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> OK, I am not extremely concerned with getting an ATA working >> right now. If I could just get an inbound extension-to-email >> working it would be great. I'll start from scratch and then let >> you know what happens. Just to make sure, I am going to create an >> extension, assign an email address to it, make sure there is an >> inbound route to that extension, and then test it. >> >> >> Robert >> >> >> ------------------------------------------------------------------------ >> Date: Wed, 16 Jan 2013 12:11:34 -0500 >> From: gassaad at emassembly.com >> To: freeswitch-users at lists.freeswitch.org >> >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> The reason for outbound extension is: >> When you send a fax from ATA to the outside world, it will dial >> first to the outbound extension and then after successful receive >> it will dial out to the intended number. >> >> As for the error you have it may be related to your ATA. I am not >> sure please see why do you get the following message. >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " >> >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig >> > wrote: >> >> OK, I've already tried creating one extension for inbound and >> set an email address. Of course, it failed and the errors are >> at the bottom of this email. What is the purpose of the >> outbound extension? >> >> Robert >> >> >> ------------------------------------------------------------------------ >> Date: Wed, 16 Jan 2013 09:57:51 -0500 >> From: gassaad at emassembly.com >> >> To: freeswitch-users at lists.freeswitch.org >> >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Sorry for the delay to get back to you. >> 1- App --> fax server >> 2- Create two extensions (one for inbound) another for >> (outbound) From within either extension you can send faxes >> "virtual fax" >> 3- point your create your outbound and inbound dialplans as >> normal but make it to point to those extensions. >> >> Good luck, >> >> George >> >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig >> > >> wrote: >> >> What did you do on the config side to get it to work? >> I've tried both an Ata and a fax to email extension and >> they both fail in both directions. >> >> Robert >> >> On Jan 12, 2013, at 2:04 PM, "George Assaad" >> > >> wrote: >> >> I use Fusion successfully to send / receive fax. >> The way it works: >> ATA --->FaxServer. >> After successful receipt, Fax Server---> DID where >> you have dialled originally. >> >> Cheers, >> >> George >> >> >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig >> > > wrote: >> >> Did anyone see the issue or what I may have >> missed in the config? As mentioned earlier, I >> thought it would after a fresh install. Any help >> is appreciated. >> >> Robert >> >> >> ------------------------------------------------------------------------ >> From: craig.robert at hotmail.com >> >> To: freeswitch-users at lists.freeswitch.org >> >> Date: Thu, 10 Jan 2013 13:43:09 -0700 >> >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Thanks for the quick response. Below is the >> capture from the log viewer. In this test, I am >> using a remote fax machine (line from Cox with >> physical machine) faxing into the "email server" >> where I created an extension for fax-to-email. >> Calling number is 1479-246-0547. >> >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:569 >> (sofia/external/+14792460547 at flowroute.com >> ) State DESTROY going to sleep >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:99 >> sofia/external/+14792460547 at flowroute.com >> Standard DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] >> mod_sofia.c:396 >> sofia/external/+14792460547 at flowroute.com >> SOFIA DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:569 >> (sofia/external/+14792460547 at flowroute.com >> ) State DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:559 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State Change >> CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:556 >> (sofia/external/+14792460547 at flowroute.com >> ) Callstate Change HANGUP >> -> DOWN >> 2013-01-10 13:39:53.162506 [NOTICE] >> switch_core_session.c:1510 Close Channel >> sofia/external/+14792460547 at flowroute.com >> [CS_DESTROY] >> 2013-01-10 13:39:53.162506 [NOTICE] >> switch_core_session.c:1506 Session 26 >> (sofia/external/+14792460547 at flowroute.com >> ) Ended >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_session.c:1488 Session 26 >> (sofia/external/+14792460547 at flowroute.com >> ) Locked, Waiting on >> external entities >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:440 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change CS_REPORTING >> -> CS_DESTROY >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:749 >> (sofia/external/+14792460547 at flowroute.com >> ) State >> REPORTING going to sleep >> 2013-01-10 13:39:53.162506 [DEBUG] >> switch_core_state_machine.c:92 >> sofia/external/+14792460547 at flowroute.com >> Standard >> REPORTING, cause: NORMAL_CLEARING >> 2013-01-10 13:39:53.066135 [DEBUG] >> switch_core_state_machine.c:749 >> (sofia/external/+14792460547 at flowroute.com >> ) State REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State >> Change CS_REPORTING >> 2013-01-10 13:39:53.066135 [DEBUG] >> switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:53.066135 [DEBUG] >> switch_core_state_machine.c:446 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change >> CS_HANGUP -> CS_REPORTING >> >> 2013-01-10 13:39:53.066135 [DEBUG] >> switch_core_state_machine.c:599 Hangup Command >> with no Session system(/usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php >> email=robert.l.craig at craignetwork.com >> >> extension=3150 >> name=+14792460547-2013-01-10-13-39-26 >> messages='result: Disconnected after permitted >> retries sender:ATW-4792460547 pages:0' >> domain=192.168.4.15 caller_id_name='' >> caller_id_number=): >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_state_machine.c:667 >> (sofia/external/+14792460547 at flowroute.com >> ) State HANGUP >> going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_state_machine.c:48 >> sofia/external/+14792460547 at flowroute.com >> Standard >> HANGUP, cause: NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_sofia.c:552 Sending BYE to >> sofia/external/+14792460547 at flowroute.com >> >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_sofia.c:503 Channel >> sofia/external/+14792460547 at flowroute.com >> hanging up, >> cause: NORMAL_CLEARING >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_state_machine.c:667 >> (sofia/external/+14792460547 at flowroute.com >> ) State HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State >> Change CS_HANGUP >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com >> ) State EXECUTE >> going to sleep >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_session.c:2687 >> sofia/external/+14792460547 at flowroute.com >> skip receive >> message [APPLICATION_EXEC_COMPLETE] (channel is >> hungup already) >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_channel.c:3017 Send signal >> sofia/external/+14792460547 at flowroute.com >> [KILL] >> 2013-01-10 13:39:52.216151 [NOTICE] >> mod_dptools.c:1150 Hangup >> sofia/external/+14792460547 at flowroute.com >> [CS_EXECUTE] >> [NORMAL_CLEARING] >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_channel.c:2994 >> (sofia/external/+14792460547 at flowroute.com >> ) Callstate >> Change ACTIVE -> HANGUP >> EXECUTE sofia/external/+14792460547 at flowroute.com >> hangup() >> 2013-01-10 13:39:52.216151 [DEBUG] >> switch_core_codec.c:244 >> sofia/external/+14792460547 at flowroute.com >> Restore >> previous codec PCMU:0. >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:518 >> ============================================================================== >> >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:516 remote model: >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:515 remote vendor: >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:514 remote country: >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:513 ECM status off >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:511 Transfer Rate: 14400 >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:510 Image resolution: 0x0 >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:509 Total fax pages: 0 >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:507 Pages transferred: 0 >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:506 Local station id: SpanDSP >> Fax Ident >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:505 Remote station id: >> ATW-4792460547 >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:500 Fax processing not >> successful - result (48) Disconnected after >> permitted retries. >> 2013-01-10 13:39:52.216151 [DEBUG] >> mod_spandsp_fax.c:487 >> ============================================================================== >> >> 2013-01-10 13:39:38.766190 [DEBUG] >> switch_rtp.c:3624 Correct ip/port confirmed. >> 2013-01-10 13:39:29.576127 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822782 IN IP4 >> 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] >> sofia_glue.c:173 >> sofia/external/+14792460547 at flowroute.com >> image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] >> mod_sofia.c:1487 Remote address:port >> [66.42.121.5:31632 ] >> has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] >> mod_sofia.c:1487 Remote address:port >> [66.42.121.5:31632 ] >> has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822781 IN IP4 >> 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.576127 [DEBUG] >> sofia_glue.c:173 >> sofia/external/+14792460547 at flowroute.com >> image media sdp: >> 2013-01-10 13:39:29.576127 [DEBUG] >> mod_sofia.c:1487 Remote address:port >> [66.42.121.5:31632 ] >> has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] >> mod_sofia.c:1487 Remote address:port >> [66.42.121.5:31632 ] >> has not changed. >> 2013-01-10 13:39:29.576127 [DEBUG] >> sofia_glue.c:3645 >> sofia/external/+14792460547 at flowroute.com >> Set rtp dtmf >> delay to 40 >> 2013-01-10 13:39:29.576127 [DEBUG] >> sofia_glue.c:3618 Set 2833 dtmf receive payload >> to 101 >> 2013-01-10 13:39:29.576127 [DEBUG] >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 >> 2013-01-10 13:39:29.576127 [ERR] >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] >> must be between 20 and 10000 >> 2013-01-10 13:39:29.576127 [DEBUG] >> switch_rtp.c:1928 Starting timer [soft] 160 bytes >> per 20ms >> 2013-01-10 13:39:29.576127 [DEBUG] >> sofia_glue.c:3348 AUDIO RTP >> [sofia/external/+14792460547 at flowroute.com >> ] 192.168.4.15 >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 >> Channel sofia/external/+14792460547 at flowroute.com >> entering state >> [ready][200] >> 2013-01-10 13:39:29.566140 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:29.566140 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> >> a=T38FaxRateManagement:transferredTCF >> m=image 31632 udptl t38 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849876 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 >> Remote SDP: >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 >> Channel sofia/external/+14792460547 at flowroute.com >> entering state >> [completing][200] >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 >> sofia/external/+14792460547 at flowroute.com >> Update Callee >> ID to "+14792460547" <+14792460547> >> 2013-01-10 13:39:29.556130 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:29.556130 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 >> Channel sofia/external/+14792460547 at flowroute.com >> entering state >> [calling][0] >> 2013-01-10 13:39:29.256196 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 >> 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] >> sofia_glue.c:2647 Local SDP: >> >> a=T38FaxUdpEC:t38UDPRedundancy >> a=T38FaxMaxDatagram:400 >> a=T38FaxMaxBuffer:2000 >> a=T38FaxRateManagement:transferredTCF >> a=T38FaxFillBitRemoval >> a=T38MaxBitRate:9600 >> a=T38FaxVersion:0 >> m=image 27588 udptl t38 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822780 IN IP4 >> 72.201.112.131 >> v=0 >> 2013-01-10 13:39:29.256196 [DEBUG] >> sofia_glue.c:173 >> sofia/external/+14792460547 at flowroute.com >> image media sdp: >> 2013-01-10 13:39:27.996269 [DEBUG] >> mod_spandsp_fax.c:1379 Raw write codec activation >> Success L16 >> 2013-01-10 13:39:27.996269 [DEBUG] >> switch_core_codec.c:219 >> sofia/external/+14792460547 at flowroute.com >> Push codec L16:70 >> 2013-01-10 13:39:27.996269 [DEBUG] >> mod_spandsp_fax.c:1363 Raw read codec activation >> Success L16 20000 >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) >> >> 2013-01-10 13:39:27.996269 [DEBUG] >> switch_ivr_play_say.c:1682 done playing file >> silence_stream://2000 >> 2013-01-10 13:39:26.256197 [DEBUG] >> switch_rtp.c:3624 Correct ip/port confirmed. >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 >> Channel sofia/external/+14792460547 at flowroute.com >> entering state >> [ready][200] >> 2013-01-10 13:39:26.096329 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:26.096329 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:26.016177 [DEBUG] >> switch_ivr_play_say.c:1309 Codec Activated >> L16 at 8000hz 1 channels 20ms >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> playback(silence_stream://2000) >> 2013-01-10 13:39:26.016177 [DEBUG] >> mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET >> [last_fax]=[+14792460547-2013-01-10-13-39-26] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> set(last_fax=+14792460547-2013-01-10-13-39-26) >> 2013-01-10 13:39:26.016177 [DEBUG] >> mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET >> [fax_enable_t38_request]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> set(fax_enable_t38_request=true) >> 2013-01-10 13:39:26.006127 [DEBUG] >> mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET >> [fax_enable_t38]=[true] >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 >> Channel sofia/external/+14792460547 at flowroute.com >> entering state >> [completed][200] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> set(fax_enable_t38=true) >> 2013-01-10 13:39:26.006127 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:26.006127 [NOTICE] >> mod_dptools.c:1176 Channel >> [sofia/external/+14792460547 at flowroute.com >> ] has been >> answered >> 2013-01-10 13:39:26.006127 [DEBUG] >> switch_channel.c:3395 >> (sofia/external/+14792460547 at flowroute.com >> ) Callstate >> Change RINGING -> ACTIVE >> 2013-01-10 13:39:26.006127 [DEBUG] >> switch_core_session.c:830 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> >> a=sendrecv >> a=ptime:20 >> a=silenceSupp:off - - - - >> a=fmtp:101 0-16 >> a=rtpmap:101 telephone-event/8000 >> a=rtpmap:0 PCMU/8000 >> m=audio 27588 RTP/AVP 0 101 >> t=0 0 >> c=IN IP4 72.201.112.131 >> s=FreeSWITCH >> o=FreeSWITCH 1357822778 1357822779 IN IP4 >> 72.201.112.131 >> v=0 >> 2013-01-10 13:39:26.006127 [DEBUG] >> mod_sofia.c:856 Local SDP >> sofia/external/+14792460547 at flowroute.com >> : >> 2013-01-10 13:39:26.006127 [DEBUG] >> sofia_glue.c:3645 >> sofia/external/+14792460547 at flowroute.com >> Set rtp dtmf >> delay to 40 >> 2013-01-10 13:39:26.006127 [DEBUG] >> sofia_glue.c:3618 Set 2833 dtmf receive payload >> to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 >> 2013-01-10 13:39:26.006127 [DEBUG] >> switch_rtp.c:1928 Starting timer [soft] 160 bytes >> per 20ms >> 2013-01-10 13:39:26.006127 [DEBUG] >> sofia_glue.c:3348 AUDIO RTP >> [sofia/external/+14792460547 at flowroute.com >> ] 192.168.4.15 >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 >> 2013-01-10 13:39:26.006127 [DEBUG] >> sofia_glue.c:1046 STUN Success >> [72.201.112.131]:[27588] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> answer() >> 2013-01-10 13:39:25.816280 [DEBUG] >> mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET >> [api_hangup_hook]=[system /usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php >> email=robert.l.craig at craignetwork.com >> >> extension=3150 name=${last_fax} messages='result: >> ${fax_result_text} >> sender:${fax_remote_station_id} >> pages:${fax_document_total_pages}' >> domain=192.168.4.15 >> caller_id_name='${caller_id_name}' >> caller_id_number=${caller_id_number}] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> set(api_hangup_hook=system /usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php >> email=robert.l.craig at craignetwork.com >> >> extension=3150 name=\${last_fax} >> messages='result: \${fax_result_text} >> sender:\${fax_remote_station_id} >> pages:\${fax_document_total_pages}' >> domain=192.168.4.15 >> caller_id_name='\${caller_id_name}' >> caller_id_number=\${caller_id_number}) >> 2013-01-10 13:39:25.816280 [DEBUG] >> mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) >> >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) >> >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) >> >> 2013-01-10 13:39:25.816280 [DEBUG] >> mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET >> [call_direction]=[local] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> set(call_direction=local) >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:209 >> sofia/external/+14792460547 at flowroute.com >> Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> mod_sofia.c:242 >> sofia/external/+14792460547 at flowroute.com >> SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com >> ) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State >> Change CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com >> ) State ROUTING >> going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:167 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change >> CS_ROUTING -> CS_EXECUTE >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action hangup() >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) >> >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> playback(silence_stream://2000) >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) >> >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> set(fax_enable_t38_request=true) >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> set(fax_enable_t38=true) >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action answer() >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> set(api_hangup_hook=system /usr/bin/php >> /var/www/html/fusionpbx/secure/fax_to_email.php >> email=robert.l.craig at craignetwork.com >> >> extension=3150 name=\\\${last_fax} >> messages='result: \\\${fax_result_text} >> sender:\\\${fax_remote_station_id} >> pages:\\\${fax_document_total_pages}' >> domain=192.168.4.15 >> caller_id_name='\\\${caller_id_name}' >> caller_id_number=\\\${caller_id_number}) >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (PASS) [Test Fax] destination_number(3150) =~ >> /^3150$/ break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->Test Fax] continue=false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [vmain_user] destination_number(3150) =~ >> /^\*97$/ break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->vmain_user] continue=false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [vmain] destination_number(3150) =~ >> /^vmain$|^\*4000$|^\*98$/ break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->vmain] continue=false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [send_to_voicemail] >> destination_number(3150) =~ /^\*99(\d{2,7})$/ >> break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->send_to_voicemail] continue=false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [extension-intercom] >> destination_number(3150) =~ /^\*8(\d{2,7})$/ >> break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->extension-intercom] continue=false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [intercept-ext] destination_number(3150) >> =~ /^\*\*(\d+)$/ break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->intercept-ext] continue=false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [call_return] destination_number(3150) =~ >> /^\*69$|^lcr$/ break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->call_return] continue=false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [call_privacy] destination_number(3150) =~ >> /^\*67(\d+)$/ break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->call_privacy] continue=false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> hash(insert/${domain_name}-last_dial/global/${uuid}) >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) >> >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) >> >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Absolute >> Condition [global] >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ >> break=never >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->global] continue=true >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [redial] destination_number(3150) =~ >> /^(redial|\*870)$/ break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->redial] continue=false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [group-intercept] destination_number(3150) >> =~ /^\*8$/ break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->group-intercept] continue=false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> ANTI-Action >> set(call_direction=local) >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [call-direction] ${call_direction}() =~ >> /^(inbound|outbound|local)$/ break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->call-direction] continue=true >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [Flowroute.11d] destination_number(3150) >> =~ /^\+?(\d{11})$/ break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [default->Flowroute.11d] continue=false >> 2013-01-10 13:39:25.816280 [INFO] >> mod_dialplan_xml.c:498 Processing +14792460547 >> <+14792460547>->3150 in context default >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:117 >> sofia/external/+14792460547 at flowroute.com >> Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] >> mod_sofia.c:149 >> sofia/external/+14792460547 at flowroute.com >> SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com >> ) State ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State >> Change CS_ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com >> ) State EXECUTE >> going to sleep >> 2013-01-10 13:39:25.816280 [NOTICE] >> switch_ivr.c:1781 Transfer >> sofia/external/+14792460547 at flowroute.com >> to >> XML[3150 at default] >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_session.c:830 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_ivr.c:1774 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change >> CS_EXECUTE -> CS_ROUTING >> EXECUTE sofia/external/+14792460547 at flowroute.com >> transfer(3150 >> XML default) >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_channel.c:1135 EXPORT (export_vars) >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) >> 2013-01-10 13:39:25.816280 [DEBUG] >> mod_dptools.c:1344 >> sofia/external/+14792460547 at flowroute.com >> SET >> [outside_call]=[true] >> EXECUTE sofia/external/+14792460547 at flowroute.com >> >> set(outside_call=true) >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:209 >> sofia/external/+14792460547 at flowroute.com >> Standard EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> mod_sofia.c:242 >> sofia/external/+14792460547 at flowroute.com >> SOFIA EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:477 >> (sofia/external/+14792460547 at flowroute.com >> ) State EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State >> Change CS_EXECUTE >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com >> ) State ROUTING >> going to sleep >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:167 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change >> CS_ROUTING -> CS_EXECUTE >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> transfer(3150 XML default) >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (PASS) [Test_Number] >> destination_number(15203039810) =~ >> /^15203039810$/ break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (PASS) [Test_Number] context(public) =~ /public/ >> break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [public->Test_Number] continue=false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [call_debug] ${call_debug}(false) =~ >> /^true$/ break=never >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [public->call_debug] continue=true >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Action >> set(outside_call=true) >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Absolute >> Condition [outside_call] >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [public->outside_call] continue=true >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ >> break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> Regex >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ >> break=on-false >> Dialplan: >> sofia/external/+14792460547 at flowroute.com >> parsing >> [public->unloop] continue=false >> 2013-01-10 13:39:25.816280 [INFO] >> mod_dialplan_xml.c:498 Processing +14792460547 >> <+14792460547>->15203039810 in context public >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:117 >> sofia/external/+14792460547 at flowroute.com >> Standard ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] >> mod_sofia.c:149 >> sofia/external/+14792460547 at flowroute.com >> SOFIA ROUTING >> 2013-01-10 13:39:25.816280 [DEBUG] >> switch_core_state_machine.c:470 >> (sofia/external/+14792460547 at flowroute.com >> ) State ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_channel.c:2003 >> (sofia/external/+14792460547 at flowroute.com >> ) Callstate >> Change DOWN -> RINGING >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State >> Change CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_core_state_machine.c:454 >> (sofia/external/+14792460547 at flowroute.com >> ) State INIT >> going to sleep >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] >> mod_sofia.c:126 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change >> CS_INIT -> CS_ROUTING >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 >> sofia/external/+14792460547 at flowroute.com >> SOFIA INIT >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_core_state_machine.c:454 >> (sofia/external/+14792460547 at flowroute.com >> ) State INIT >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State >> Change CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_core_session.c:1283 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 >> (sofia/external/+14792460547 at flowroute.com >> ) State Change >> CS_NEW -> CS_INIT >> 2013-01-10 13:39:25.806120 [DEBUG] >> sofia_glue.c:5263 Set 2833 dtmf send/recv payload >> to 101 >> 2013-01-10 13:39:25.806120 [DEBUG] >> switch_core_codec.c:111 >> sofia/external/+14792460547 at flowroute.com >> Original read >> codec set to PCMU:0 >> 2013-01-10 13:39:25.806120 [DEBUG] >> sofia_glue.c:3093 Set Codec >> sofia/external/+14792460547 at flowroute.com >> PCMU/8000 20 >> ms 160 samples 64000 bits >> 2013-01-10 13:39:25.806120 [DEBUG] >> sofia_glue.c:5134 Audio Codec Compare >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >> 2013-01-10 13:39:25.806120 [DEBUG] >> sofia_glue.c:5134 Audio Codec Compare >> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >> >> a=ptime:20 >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:18 G729/8000 >> m=audio 31632 RTP/AVP 0 18 101 >> t=0 0 >> c=IN IP4 66.42.121.5 >> s=- >> o=- 275426305 1357849875 IN IP4 66.42.121.5 >> v=0 >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 >> Remote SDP: >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 >> Channel sofia/external/+14792460547 at flowroute.com >> entering state >> [received][100] >> 2013-01-10 13:39:25.786370 [DEBUG] >> switch_core_state_machine.c:433 >> (sofia/external/+14792460547 at flowroute.com >> ) State NEW >> 2013-01-10 13:39:25.786370 [DEBUG] >> switch_core_state_machine.c:415 >> (sofia/external/+14792460547 at flowroute.com >> ) Running State >> Change CS_NEW >> 2013-01-10 13:39:25.786370 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.786370 [DEBUG] >> switch_core_session.c:975 Send signal >> sofia/external/+14792460547 at flowroute.com >> [BREAK] >> 2013-01-10 13:39:25.786370 [NOTICE] >> switch_channel.c:968 New Channel >> sofia/external/+14792460547 at flowroute.com >> >> [d276f012-5b65-11e2-beae-8564ff827a25] >> >> Robert >> >> >> ------------------------------------------------------------------------ >> Date: Thu, 10 Jan 2013 09:47:18 -0800 >> From: msc at freeswitch.org >> To: freeswitch-users at lists.freeswitch.org >> >> Subject: Re: [Freeswitch-users] T38 Configuration >> >> Can you pastebin a console debug log w/ sip trace >> of a call failing to negotiate t.38 correctly? >> That might help us figure out where to look. >> -MC >> >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig >> > > wrote: >> >> Just installed FusionPBX using the script >> built for a Centos 6 install. Everything >> works great in regards to calls, but my main >> goal of this project was to utilize the fax >> capabilities of Freeswitch/FusionPBX. I have >> tried many things in media bypass, routing to >> a fax extension, using an ATA, etc., but I >> can't get T38 to negotiate correctly. Can >> someone point me in the right direction as to >> where I should begin? I am guessing it isn't >> supposed to work right out of the box. I am >> using Flowroute as my provider right now and >> the have confirmed that they utilize T38 in >> both directions. Thanks! >> >> >> Robert >> From steveayre at gmail.com Fri Jan 18 19:19:40 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Jan 2013 16:19:40 +0000 Subject: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 In-Reply-To: <50F91490.7080006@gmail.com> References: <50F909AD.2010308@gmail.com> <50F91490.7080006@gmail.com> Message-ID: First of all update to the latest version of v1.2.stable or master - it may be something that's already fixed if you're on an old version. Second, configure FreeSWITCH to save a coredump on a segfault (start it from the init.d script with the -core param). Once you have a coredump you can open a Jira ticket with a backtrace: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 The backtrace will show exactly where and why the segfault occurred, very helpful for debugging. Any additional info eg FS debug logs would also be helpful. -Steve On 18 January 2013 09:23, Mimiko wrote: > New one: > kernel: freeswitch[3559]: segfault at 30 ip 00007fbb945862a4 sp > 00007fbb8ef59bc0 error 4 in libfreeswitch.so.1.0.0[7fbb944fb000+277000] > > This happened when channels where destroyed after a bridge and a hangup. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/76e071e1/attachment.html From craig.robert at hotmail.com Fri Jan 18 19:26:32 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Fri, 18 Jan 2013 09:26:32 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: <50F975E5.2040206@coppice.org> References: , , , , , , , , , , , , , <50F975E5.2040206@coppice.org> Message-ID: Where does the file get downloaded to? What should the permissions be? Robert > Date: Sat, 19 Jan 2013 00:18:45 +0800 > From: steveu at coppice.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If the TIFF file cannot be opened on receive you probably have a file > permissions issue (e.g. with the directory in which the file is to be > created). If the TIFF file cannot be opened on transmit there are > various possibilities about the suitability of the file's contents. > > Steve > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > This was an incoming fax. I don't have much control over the other > > end. Why is that error appearing when trying to receive something? > > > > Robert > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > wrote: > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > >> Please try pdf, you should be in business now. > >> Please let us know, > >> > >> On Jan 18, 2013 12:35 AM, "Robert Craig" >> > wrote: > >> > >> OK, so I've made a little bit of progress. The distant end is a > >> Windows Fax server hooked up to an analog line. I created a fax > >> extension in FS, assigned an inbound route to the extension, and > >> made sure the extension had an email address. I believe the fax > >> receive wants to complete, but there is an interesting error in > >> the log about TIF files. Anyone ever seen this before? > >> > >> v=0 > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> c=IN IP4 72.201.112.131 > >> t=0 0 > >> m=image 27714 udptl t38 > >> a=T38FaxVersion:0 > >> a=T38MaxBitRate:9600 > >> a=T38FaxFillBitRemoval > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxUdpEC:t38UDPRedundancy > >> > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > >> ip/port confirmed. > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > >> Negotiation Result > >> ======================================================= > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > >> status off > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > >> country: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > >> vendor: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > >> model: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > >> processing not successful - result (41) TIFF/F file cannot be > >> opened.* > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > >> transferred: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > >> fax pages: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > >> resolution: 0x0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > >> status off > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > >> country: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > >> vendor: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > >> model: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > >> sofia/external/+14792460547 @flowroute.com > >> Restore previous codec PCMU:0. > >> EXECUTE sofia/external/+14792460547 > >> @flowroute.com hangup() > >> > >> Robert > >> > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> OK, I am not extremely concerned with getting an ATA working > >> right now. If I could just get an inbound extension-to-email > >> working it would be great. I'll start from scratch and then let > >> you know what happens. Just to make sure, I am going to create an > >> extension, assign an email address to it, make sure there is an > >> inbound route to that extension, and then test it. > >> > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > >> From: gassaad at emassembly.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> The reason for outbound extension is: > >> When you send a fax from ATA to the outside world, it will dial > >> first to the outbound extension and then after successful receive > >> it will dial out to the intended number. > >> > >> As for the error you have it may be related to your ATA. I am not > >> sure please see why do you get the following message. > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > >> > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > >> > wrote: > >> > >> OK, I've already tried creating one extension for inbound and > >> set an email address. Of course, it failed and the errors are > >> at the bottom of this email. What is the purpose of the > >> outbound extension? > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > >> From: gassaad at emassembly.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Sorry for the delay to get back to you. > >> 1- App --> fax server > >> 2- Create two extensions (one for inbound) another for > >> (outbound) From within either extension you can send faxes > >> "virtual fax" > >> 3- point your create your outbound and inbound dialplans as > >> normal but make it to point to those extensions. > >> > >> Good luck, > >> > >> George > >> > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > >> > > >> wrote: > >> > >> What did you do on the config side to get it to work? > >> I've tried both an Ata and a fax to email extension and > >> they both fail in both directions. > >> > >> Robert > >> > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > >> > > >> wrote: > >> > >> I use Fusion successfully to send / receive fax. > >> The way it works: > >> ATA --->FaxServer. > >> After successful receipt, Fax Server---> DID where > >> you have dialled originally. > >> > >> Cheers, > >> > >> George > >> > >> > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > >> >> > wrote: > >> > >> Did anyone see the issue or what I may have > >> missed in the config? As mentioned earlier, I > >> thought it would after a fresh install. Any help > >> is appreciated. > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Thanks for the quick response. Below is the > >> capture from the log viewer. In this test, I am > >> using a remote fax machine (line from Cox with > >> physical machine) faxing into the "email server" > >> where I created an extension for fax-to-email. > >> Calling number is 1479-246-0547. > >> > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:99 > >> sofia/external/+14792460547 at flowroute.com > >> Standard DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> mod_sofia.c:396 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:559 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State Change > >> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:556 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate Change HANGUP > >> -> DOWN > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1510 Close Channel > >> sofia/external/+14792460547 at flowroute.com > >> [CS_DESTROY] > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1506 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Ended > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1488 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Locked, Waiting on > >> external entities > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:440 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change CS_REPORTING > >> -> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State > >> REPORTING going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:92 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> REPORTING, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:446 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_HANGUP -> CS_REPORTING > >> > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:599 Hangup Command > >> with no Session system(/usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 > >> name=+14792460547-2013-01-10-13-39-26 > >> messages='result: Disconnected after permitted > >> retries sender:ATW-4792460547 pages:0' > >> domain=192.168.4.15 caller_id_name='' > >> caller_id_number=): > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:48 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> HANGUP, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:552 Sending BYE to > >> sofia/external/+14792460547 at flowroute.com > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:503 Channel > >> sofia/external/+14792460547 at flowroute.com > >> hanging up, > >> cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:2687 > >> sofia/external/+14792460547 at flowroute.com > >> skip receive > >> message [APPLICATION_EXEC_COMPLETE] (channel is > >> hungup already) > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:3017 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [KILL] > >> 2013-01-10 13:39:52.216151 [NOTICE] > >> mod_dptools.c:1150 Hangup > >> sofia/external/+14792460547 at flowroute.com > >> [CS_EXECUTE] > >> [NORMAL_CLEARING] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:2994 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change ACTIVE -> HANGUP > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> hangup() > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_codec.c:244 > >> sofia/external/+14792460547 at flowroute.com > >> Restore > >> previous codec PCMU:0. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:516 remote model: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:515 remote vendor: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:514 remote country: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:513 ECM status off > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:509 Total fax pages: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:507 Pages transferred: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > >> Fax Ident > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:505 Remote station id: > >> ATW-4792460547 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:500 Fax processing not > >> successful - result (48) Disconnected after > >> permitted retries. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> 2013-01-10 13:39:38.766190 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:29.576127 [ERR] > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > >> must be between 20 and 10000 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxRateManagement:transferredTCF > >> m=image 31632 udptl t38 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completing][200] > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > >> sofia/external/+14792460547 at flowroute.com > >> Update Callee > >> ID to "+14792460547" <+14792460547> > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [calling][0] > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:2647 Local SDP: > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1379 Raw write codec activation > >> Success L16 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_core_codec.c:219 > >> sofia/external/+14792460547 at flowroute.com > >> Push codec L16:70 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1363 Raw read codec activation > >> Success L16 20000 > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > >> > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_ivr_play_say.c:1682 done playing file > >> silence_stream://2000 > >> 2013-01-10 13:39:26.256197 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> switch_ivr_play_say.c:1309 Codec Activated > >> L16 at 8000hz 1 channels 20ms > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> playback(silence_stream://2000) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38_request]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38_request=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38]=[true] > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completed][200] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.006127 [NOTICE] > >> mod_dptools.c:1176 Channel > >> [sofia/external/+14792460547 at flowroute.com > >> ] has been > >> answered > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_channel.c:3395 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change RINGING -> ACTIVE > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=sendrecv > >> a=ptime:20 > >> a=silenceSupp:off - - - - > >> a=fmtp:101 0-16 > >> a=rtpmap:101 telephone-event/8000 > >> a=rtpmap:0 PCMU/8000 > >> m=audio 27588 RTP/AVP 0 101 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_sofia.c:856 Local SDP > >> sofia/external/+14792460547 at flowroute.com > >> : > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:1046 STUN Success > >> [72.201.112.131]:[27588] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> answer() > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [api_hangup_hook]=[system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=${last_fax} messages='result: > >> ${fax_result_text} > >> sender:${fax_remote_station_id} > >> pages:${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='${caller_id_name}' > >> caller_id_number=${caller_id_number}] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\${last_fax} > >> messages='result: \${fax_result_text} > >> sender:\${fax_remote_station_id} > >> pages:\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\${caller_id_name}' > >> caller_id_number=\${caller_id_number}) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [call_direction]=[local] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(call_direction=local) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action hangup() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> playback(silence_stream://2000) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38_request=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action answer() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\\\${last_fax} > >> messages='result: \\\${fax_result_text} > >> sender:\\\${fax_remote_station_id} > >> pages:\\\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\\\${caller_id_name}' > >> caller_id_number=\\\${caller_id_number}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test Fax] destination_number(3150) =~ > >> /^3150$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Test Fax] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain_user] destination_number(3150) =~ > >> /^\*97$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain_user] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain] destination_number(3150) =~ > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [send_to_voicemail] > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->send_to_voicemail] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [extension-intercom] > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->extension-intercom] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [intercept-ext] destination_number(3150) > >> =~ /^\*\*(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->intercept-ext] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_return] destination_number(3150) =~ > >> /^\*69$|^lcr$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_return] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_privacy] destination_number(3150) =~ > >> /^\*67(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_privacy] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [global] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > >> break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->global] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [redial] destination_number(3150) =~ > >> /^(redial|\*870)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->redial] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [group-intercept] destination_number(3150) > >> =~ /^\*8$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->group-intercept] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> ANTI-Action > >> set(call_direction=local) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call-direction] ${call_direction}() =~ > >> /^(inbound|outbound|local)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call-direction] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [Flowroute.11d] destination_number(3150) > >> =~ /^\+?(\d{11})$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Flowroute.11d] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->3150 in context default > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:25.816280 [NOTICE] > >> switch_ivr.c:1781 Transfer > >> sofia/external/+14792460547 at flowroute.com > >> to > >> XML[3150 at default] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_ivr.c:1774 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_EXECUTE -> CS_ROUTING > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> transfer(3150 > >> XML default) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_channel.c:1135 EXPORT (export_vars) > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [outside_call]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(outside_call=true) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> transfer(3150 XML default) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] > >> destination_number(15203039810) =~ > >> /^15203039810$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] context(public) =~ /public/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->Test_Number] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_debug] ${call_debug}(false) =~ > >> /^true$/ break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->call_debug] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(outside_call=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [outside_call] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->outside_call] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->unloop] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->15203039810 in context public > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_channel.c:2003 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change DOWN -> RINGING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:454 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State INIT > >> going to sleep > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> mod_sofia.c:126 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_INIT -> CS_ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:454 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_NEW -> CS_INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5263 Set 2833 dtmf send/recv payload > >> to 101 > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_codec.c:111 > >> sofia/external/+14792460547 at flowroute.com > >> Original read > >> codec set to PCMU:0 > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:3093 Set Codec > >> sofia/external/+14792460547 at flowroute.com > >> PCMU/8000 20 > >> ms 160 samples 64000 bits > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5134 Audio Codec Compare > >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5134 Audio Codec Compare > >> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > >> > >> a=ptime:20 > >> a=fmtp:101 0-15 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:18 annexb=no > >> a=rtpmap:18 G729/8000 > >> m=audio 31632 RTP/AVP 0 18 101 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849875 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [received][100] > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_state_machine.c:433 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State NEW > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_NEW > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.786370 [NOTICE] > >> switch_channel.c:968 New Channel > >> sofia/external/+14792460547 at flowroute.com > >> > >> [d276f012-5b65-11e2-beae-8564ff827a25] > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Thu, 10 Jan 2013 09:47:18 -0800 > >> From: msc at freeswitch.org > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Can you pastebin a console debug log w/ sip trace > >> of a call failing to negotiate t.38 correctly? > >> That might help us figure out where to look. > >> -MC > >> > >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig > >> >> > wrote: > >> > >> Just installed FusionPBX using the script > >> built for a Centos 6 install. Everything > >> works great in regards to calls, but my main > >> goal of this project was to utilize the fax > >> capabilities of Freeswitch/FusionPBX. I have > >> tried many things in media bypass, routing to > >> a fax extension, using an ATA, etc., but I > >> can't get T38 to negotiate correctly. Can > >> someone point me in the right direction as to > >> where I should begin? I am guessing it isn't > >> supposed to work right out of the box. I am > >> using Flowroute as my provider right now and > >> the have confirmed that they utilize T38 in > >> both directions. Thanks! > >> > >> > >> Robert > >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/c6290ebf/attachment-0001.html From admin at smallunix.net Fri Jan 18 19:30:02 2013 From: admin at smallunix.net (Andrea Mazzeo) Date: Fri, 18 Jan 2013 17:30:02 +0100 Subject: [Freeswitch-users] SONUS - dtg inside INVITE In-Reply-To: References: <7443520999689776200@unknownmsgid> <5D31EF90-20B7-48BA-B7BC-65F30B5383DD@jerris.com> Message-ID: The provider is Switchover 2013/1/18 Kristian Kielhofner : > Very interesting... > > If you don't mind saying, what provider is this? > > On Fri, Jan 18, 2013 at 9:46 AM, Andrea Mazzeo wrote: >> Hi Kristian, >> >> I did a trace, and it's in the Contact URI. >> >> >> 2013/1/18 Kristian Kielhofner : >>> Are you sure about that? >>> >>> 99% of the time this stuff goes in the destination URI (either the >>> host or user portion), which leads back to my original question... >>> >>> On Fri, Jan 18, 2013 at 3:01 AM, Andrea Mazzeo wrote: >>>> Hi Michael, >>>> >>>> In the contact field. >>>> >>>> 2013/1/18 Michael Jerris : >>>>> What you posted is NOT a valid invite header. WHERE in the invite header do you need it? >>>>> >>>>> On Jan 17, 2013, at 6:59 PM, Andrea Mazzeo wrote: >>>>> >>>>>> Hi Kristian, >>>>>> >>>>>> I need dtg in the INVITE header. >>>>>> >>>>>> Best regards, >>>>>> Andrea Mazzeo >>>>>> >>>>>> Il giorno 17/gen/2013, alle ore 22:01, Kristian Kielhofner >>>>>> ha scritto: >>>>>> >>>>>>> This is almost certainly not the syntax you want to use. >>>>>>> >>>>>>> Do that expect "dtg" to be a user or domain parameter? >>>>>>> >>>>>>> On Thu, Jan 17, 2013 at 1:46 PM, Andrea Mazzeo wrote: >>>>>>>> Hi all, >>>>>>>> >>>>>>>> I need to configure my FS to send call to a Sonus GSX via SIP. >>>>>>>> Is there a way to send SIP parameter "dtg" (destination trunk group) >>>>>>>> inside the INVITE? >>>>>>>> >>>>>>>> Example: >>>>>>>> >>>>>>>> Request-Line: INVITE sip: 39340xxxxxx at 38.xxx.xxx.xxx SIP/2.0 dtg=ATTMDT7807IPO >>>>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> -- >>> Kristian Kielhofner >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From leshadorofeev at gmail.com Fri Jan 18 16:38:55 2013 From: leshadorofeev at gmail.com (=?KOI8-R?B?4czFy9PFyiDkz9LPxsXF1w==?=) Date: Fri, 18 Jan 2013 17:38:55 +0400 Subject: [Freeswitch-users] FS hangs up after 30 seconds by timeout on ACK (after sending 200 OK) Message-ID: Hi, all! I have two sip profiles: profile_spb and profile_msk for one gateway. I assign sip port 5060 for profile_spb and port 5070 for profile_msk. FS has successfully registered two users with distinct usernames: num_spb and num_msk on the gateway. By the way, field "Contact" in message REGISTER has normal port number - 5060 and 5070 accordingly. When someone calls to num_spb, it shoot to port 5060 and all works well. When someone calls to num_msk, it shoot to port 5070. Call is received, but is rejected after 30 seconds by timeout (not received ACK on 200 OK). I think it's because FS sends 200 OK with port number = 0 in field Contact. All another headers are good (it seems). Does someone know how to say FS to use correct port number in field Contact in 200 OK? Does someone has another version? Thanks a lot! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/b189c173/attachment.html From msc at freeswitch.org Fri Jan 18 19:49:31 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Jan 2013 08:49:31 -0800 Subject: [Freeswitch-users] mod_xml_curl - request for comments In-Reply-To: <804D48104511D4468F0D60DF9D3100350B0A1102@MAIL.millicorp.com> References: <804D48104511D4468F0D60DF9D3100350B0A1102@MAIL.millicorp.com> Message-ID: Agreed, this is a significant improvement. Keep plugging away and if anyone else has the time and knowledge to add to this page that would be fantastic. -MC On Fri, Jan 18, 2013 at 6:50 AM, Tim Meade wrote: > Hi Cal?.**** > > ** ** > > I think it looks MUCH better. **** > > ** ** > > Great job,**** > > ** ** > > Tim**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal Leeming > [Simplicity Media Ltd] > *Sent:* Friday, January 18, 2013 9:33 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] mod_xml_curl - request for comments**** > > ** ** > > Hello,**** > > ** ** > > I've been re-reading my changes to mod_xml_curl documentation, and I'm > still not entirely happy with it.**** > > ** ** > > I can't tell if it genuinely needs a different approach, or if I'm just > being fussy.**** > > ** ** > > Therefore, constructive criticism is very much welcome - even if it means > the whole damn thing has to be re-done again.**** > > ** ** > > http://wiki.freeswitch.org/wiki/Mod_xml_curl**** > > ** ** > > Thanks**** > > > Cal**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/4ce92668/attachment.html From msc at freeswitch.org Fri Jan 18 20:13:36 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Jan 2013 09:13:36 -0800 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: <50F975E5.2040206@coppice.org> Message-ID: If you are using the rxfax app then you specify the file name and location: Make sure that the target directory has read/write permissions. You can use something like "chmod 666 /my/directory" to make it read/write. -MC On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: > Where does the file get downloaded to? What should the permissions be? > > Robert > > > > Date: Sat, 19 Jan 2013 00:18:45 +0800 > > From: steveu at coppice.org > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] T38 Configuration > > > > If the TIFF file cannot be opened on receive you probably have a file > > permissions issue (e.g. with the directory in which the file is to be > > created). If the TIFF file cannot be opened on transmit there are > > various possibilities about the suitability of the file's contents. > > > > Steve > > > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > > This was an incoming fax. I don't have much control over the other > > > end. Why is that error appearing when trying to receive something? > > > > > > Robert > > > > > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > > wrote: > > > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > > >> Please try pdf, you should be in business now. > > >> Please let us know, > > >> > > >> On Jan 18, 2013 12:35 AM, "Robert Craig" > >> > wrote: > > >> > > >> OK, so I've made a little bit of progress. The distant end is a > > >> Windows Fax server hooked up to an analog line. I created a fax > > >> extension in FS, assigned an inbound route to the extension, and > > >> made sure the extension had an email address. I believe the fax > > >> receive wants to complete, but there is an interesting error in > > >> the log about TIF files. Anyone ever seen this before? > > >> > > >> v=0 > > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> c=IN IP4 72.201.112.131 > > >> t=0 0 > > >> m=image 27714 udptl t38 > > >> a=T38FaxVersion:0 > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxFillBitRemoval > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > > >> ip/port confirmed. > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > > >> Negotiation Result > > >> ======================================================= > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > > >> status off > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > > >> country: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > > >> vendor: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > > >> model: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > > >> processing not successful - result (41) TIFF/F file cannot be > > >> opened.* > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > > >> transferred: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > > >> fax pages: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > > >> resolution: 0x0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > > >> status off > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > > >> country: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > > >> vendor: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > > >> model: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > > >> sofia/external/+14792460547 @flowroute.com > > >> Restore previous codec PCMU:0. > > >> EXECUTE sofia/external/+14792460547 > > >> @flowroute.com hangup() > > >> > > >> Robert > > >> > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> OK, I am not extremely concerned with getting an ATA working > > >> right now. If I could just get an inbound extension-to-email > > >> working it would be great. I'll start from scratch and then let > > >> you know what happens. Just to make sure, I am going to create an > > >> extension, assign an email address to it, make sure there is an > > >> inbound route to that extension, and then test it. > > >> > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > > >> From: gassaad at emassembly.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> The reason for outbound extension is: > > >> When you send a fax from ATA to the outside world, it will dial > > >> first to the outbound extension and then after successful receive > > >> it will dial out to the intended number. > > >> > > >> As for the error you have it may be related to your ATA. I am not > > >> sure please see why do you get the following message. > > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > > >> > > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > > >> > wrote: > > >> > > >> OK, I've already tried creating one extension for inbound and > > >> set an email address. Of course, it failed and the errors are > > >> at the bottom of this email. What is the purpose of the > > >> outbound extension? > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > > >> From: gassaad at emassembly.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Sorry for the delay to get back to you. > > >> 1- App --> fax server > > >> 2- Create two extensions (one for inbound) another for > > >> (outbound) From within either extension you can send faxes > > >> "virtual fax" > > >> 3- point your create your outbound and inbound dialplans as > > >> normal but make it to point to those extensions. > > >> > > >> Good luck, > > >> > > >> George > > >> > > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > > >> > > > >> wrote: > > >> > > >> What did you do on the config side to get it to work? > > >> I've tried both an Ata and a fax to email extension and > > >> they both fail in both directions. > > >> > > >> Robert > > >> > > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > > >> > > > >> wrote: > > >> > > >> I use Fusion successfully to send / receive fax. > > >> The way it works: > > >> ATA --->FaxServer. > > >> After successful receipt, Fax Server---> DID where > > >> you have dialled originally. > > >> > > >> Cheers, > > >> > > >> George > > >> > > >> > > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > > >> > >> > wrote: > > >> > > >> Did anyone see the issue or what I may have > > >> missed in the config? As mentioned earlier, I > > >> thought it would after a fresh install. Any help > > >> is appreciated. > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Thanks for the quick response. Below is the > > >> capture from the log viewer. In this test, I am > > >> using a remote fax machine (line from Cox with > > >> physical machine) faxing into the "email server" > > >> where I created an extension for fax-to-email. > > >> Calling number is 1479-246-0547. > > >> > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:99 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> mod_sofia.c:396 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:559 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State Change > > >> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:556 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate Change HANGUP > > >> -> DOWN > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1510 Close Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_DESTROY] > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1506 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Ended > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1488 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Locked, Waiting on > > >> external entities > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:440 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change CS_REPORTING > > >> -> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State > > >> REPORTING going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:92 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> REPORTING, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:446 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_HANGUP -> CS_REPORTING > > >> > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:599 Hangup Command > > >> with no Session system(/usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 > > >> name=+14792460547-2013-01-10-13-39-26 > > >> messages='result: Disconnected after permitted > > >> retries sender:ATW-4792460547 pages:0' > > >> domain=192.168.4.15 caller_id_name='' > > >> caller_id_number=): > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:48 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> HANGUP, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:552 Sending BYE to > > >> sofia/external/+14792460547 at flowroute.com > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:503 Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> hanging up, > > >> cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:2687 > > >> sofia/external/+14792460547 at flowroute.com > > >> skip receive > > >> message [APPLICATION_EXEC_COMPLETE] (channel is > > >> hungup already) > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:3017 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [KILL] > > >> 2013-01-10 13:39:52.216151 [NOTICE] > > >> mod_dptools.c:1150 Hangup > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_EXECUTE] > > >> [NORMAL_CLEARING] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:2994 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change ACTIVE -> HANGUP > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> hangup() > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_codec.c:244 > > >> sofia/external/+14792460547 at flowroute.com > > >> Restore > > >> previous codec PCMU:0. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:516 remote model: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:515 remote vendor: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:514 remote country: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:513 ECM status off > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:509 Total fax pages: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:507 Pages transferred: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > > >> Fax Ident > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:505 Remote station id: > > >> ATW-4792460547 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:500 Fax processing not > > >> successful - result (48) Disconnected after > > >> permitted retries. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:38.766190 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:29.576127 [ERR] > > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > > >> must be between 20 and 10000 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxRateManagement:transferredTCF > > >> m=image 31632 udptl t38 > > >> t=0 0 > > >> c=IN IP4 66.42.121.5 > > >> s=- > > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > > >> v=0 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > > >> Remote SDP: > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completing][200] > > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > > >> sofia/external/+14792460547 at flowroute.com > > >> Update Callee > > >> ID to "+14792460547" <+14792460547> > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [calling][0] > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:2647 Local SDP: > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1379 Raw write codec activation > > >> Success L16 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_core_codec.c:219 > > >> sofia/external/+14792460547 at flowroute.com > > >> Push codec L16:70 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1363 Raw read codec activation > > >> Success L16 20000 > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > > >> > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_ivr_play_say.c:1682 done playing file > > >> silence_stream://2000 > > >> 2013-01-10 13:39:26.256197 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> switch_ivr_play_say.c:1309 Codec Activated > > >> L16 at 8000hz 1 channels 20ms > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> playback(silence_stream://2000) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38_request]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38_request=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38]=[true] > > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completed][200] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.006127 [NOTICE] > > >> mod_dptools.c:1176 Channel > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] has been > > >> answered > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_channel.c:3395 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change RINGING -> ACTIVE > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=sendrecv > > >> a=ptime:20 > > >> a=silenceSupp:off - - - - > > >> a=fmtp:101 0-16 > > >> a=rtpmap:101 telephone-event/8000 > > >> a=rtpmap:0 PCMU/8000 > > >> m=audio 27588 RTP/AVP 0 101 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_sofia.c:856 Local SDP > > >> sofia/external/+14792460547 at flowroute.com > > >> : > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:1046 STUN Success > > >> [72.201.112.131]:[27588] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> answer() > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [api_hangup_hook]=[system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=${last_fax} messages='result: > > >> ${fax_result_text} > > >> sender:${fax_remote_station_id} > > >> pages:${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='${caller_id_name}' > > >> caller_id_number=${caller_id_number}] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\${last_fax} > > >> messages='result: \${fax_result_text} > > >> sender:\${fax_remote_station_id} > > >> pages:\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\${caller_id_name}' > > >> caller_id_number=\${caller_id_number}) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [call_direction]=[local] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(call_direction=local) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action hangup() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> playback(silence_stream://2000) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38_request=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action answer() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\\\${last_fax} > > >> messages='result: \\\${fax_result_text} > > >> sender:\\\${fax_remote_station_id} > > >> pages:\\\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\\\${caller_id_name}' > > >> caller_id_number=\\\${caller_id_number}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test Fax] destination_number(3150) =~ > > >> /^3150$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Test Fax] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain_user] destination_number(3150) =~ > > >> /^\*97$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain_user] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain] destination_number(3150) =~ > > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [send_to_voicemail] > > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->send_to_voicemail] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [extension-intercom] > > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->extension-intercom] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [intercept-ext] destination_number(3150) > > >> =~ /^\*\*(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->intercept-ext] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_return] destination_number(3150) =~ > > >> /^\*69$|^lcr$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_return] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_privacy] destination_number(3150) =~ > > >> /^\*67(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_privacy] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [global] > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > > >> break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->global] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [redial] destination_number(3150) =~ > > >> /^(redial|\*870)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->redial] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [group-intercept] destination_number(3150) > > >> =~ /^\*8$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->group-intercept] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> ANTI-Action > > >> set(call_direction=local) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call-direction] ${call_direction}() =~ > > >> /^(inbound|outbound|local)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call-direction] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [Flowroute.11d] destination_number(3150) > > >> =~ /^\+?(\d{11})$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Flowroute.11d] continue=false > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->3150 in context default > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [NOTICE] > > >> switch_ivr.c:1781 Transfer > > >> sofia/external/+14792460547 at flowroute.com > > >> to > > >> XML[3150 at default] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_ivr.c:1774 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_EXECUTE -> CS_ROUTING > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> transfer(3150 > > >> XML default) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_channel.c:1135 EXPORT (export_vars) > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [outside_call]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(outside_call=true) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> transfer(3150 XML default) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] > > >> destination_number(15203039810) =~ > > >> /^15203039810$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] context(public) =~ /public/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->Test_Number] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_debug] ${call_debug}(false) =~ > > >> /^true$/ break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->call_debug] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(outside_call=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [outside_call] > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->outside_call] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->unloop] continue=false > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->15203039810 in context public > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_channel.c:2003 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change DOWN -> RINGING > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_ROUTING > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:454 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State INIT > > >> going to sleep > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> mod_sofia.c:126 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_INIT -> CS_ROUTING > > >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:454 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_NEW -> CS_INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:5263 Set 2833 dtmf send/recv payload > > >> to 101 > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_codec.c:111 > > >> sofia/external/+14792460547 at flowroute.com > > >> Original read > > >> codec set to PCMU:0 > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:3093 Set Codec > > >> sofia/external/+14792460547 at flowroute.com > > >> PCMU/8000 20 > > >> ms 160 samples 64000 bits > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:5134 Audio Codec Compare > > >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:5134 Audio Codec Compare > > >> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > > >> > > >> a=ptime:20 > > >> a=fmtp:101 0-15 > > >> a=rtpmap:101 telephone-event/8000 > > >> a=fmtp:18 annexb=no > > >> a=rtpmap:18 G729/8000 > > >> m=audio 31632 RTP/AVP 0 18 101 > > >> t=0 0 > > >> c=IN IP4 66.42.121.5 > > >> s=- > > >> o=- 275426305 1357849875 IN IP4 66.42.121.5 > > >> v=0 > > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 > > >> Remote SDP: > > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [received][100] > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_state_machine.c:433 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State NEW > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_NEW > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.786370 [NOTICE] > > >> switch_channel.c:968 New Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> > > >> [d276f012-5b65-11e2-beae-8564ff827a25] > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Thu, 10 Jan 2013 09:47:18 -0800 > > >> From: msc at freeswitch.org > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Can you pastebin a console debug log w/ sip trace > > >> of a call failing to negotiate t.38 correctly? > > >> That might help us figure out where to look. > > >> -MC > > >> > > >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig > > >> > >> > wrote: > > >> > > >> Just installed FusionPBX using the script > > >> built for a Centos 6 install. Everything > > >> works great in regards to calls, but my main > > >> goal of this project was to utilize the fax > > >> capabilities of Freeswitch/FusionPBX. I have > > >> tried many things in media bypass, routing to > > >> a fax extension, using an ATA, etc., but I > > >> can't get T38 to negotiate correctly. Can > > >> someone point me in the right direction as to > > >> where I should begin? I am guessing it isn't > > >> supposed to work right out of the box. I am > > >> using Flowroute as my provider right now and > > >> the have confirmed that they utilize T38 in > > >> both directions. Thanks! > > >> > > >> > > >> Robert > > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/1e68f836/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Fri Jan 18 20:34:04 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 18 Jan 2013 17:34:04 +0000 Subject: [Freeswitch-users] mod_xml_curl - request for comments In-Reply-To: References: <804D48104511D4468F0D60DF9D3100350B0A1102@MAIL.millicorp.com> Message-ID: I agree that having some alternative input/ideas on this page would certainly be a good thing. I'm almost tempted to say that this page doesn't need to explain/detail any of the channel variable contexts (the ones where it explains Applicable common fields etc), as this is really FreeSWITCH wide, and interpretation of these variables mostly depends on your own business logic. Removing this info would make the page a lot more read-able, but at the same time, might make it more difficult to understand for new users. Perhaps a warning at the top somewhere explaining this might be more suitable? Cal On Fri, Jan 18, 2013 at 4:49 PM, Michael Collins wrote: > Agreed, this is a significant improvement. Keep plugging away and if > anyone else has the time and knowledge to add to this page that would be > fantastic. > -MC > > On Fri, Jan 18, 2013 at 6:50 AM, Tim Meade wrote: > >> Hi Cal?.**** >> >> ** ** >> >> I think it looks MUCH better. **** >> >> ** ** >> >> Great job,**** >> >> ** ** >> >> Tim**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal >> Leeming [Simplicity Media Ltd] >> *Sent:* Friday, January 18, 2013 9:33 AM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] mod_xml_curl - request for comments**** >> >> ** ** >> >> Hello,**** >> >> ** ** >> >> I've been re-reading my changes to mod_xml_curl documentation, and I'm >> still not entirely happy with it.**** >> >> ** ** >> >> I can't tell if it genuinely needs a different approach, or if I'm just >> being fussy.**** >> >> ** ** >> >> Therefore, constructive criticism is very much welcome - even if it means >> the whole damn thing has to be re-done again.**** >> >> ** ** >> >> http://wiki.freeswitch.org/wiki/Mod_xml_curl**** >> >> ** ** >> >> Thanks**** >> >> >> Cal**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/e954ad96/attachment.html From craig.robert at hotmail.com Fri Jan 18 20:43:18 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Fri, 18 Jan 2013 10:43:18 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: , , , , , , , , , , , , , <50F975E5.2040206@coppice.org>, , Message-ID: I believe I'm just using the built in SpanDSP for faxing. I've looked around for 'rxfax' and can't find anything. I did notice that the directory where the faxes were supposed to be stored for the extension I create didn't even exist. I created them, applied the permissions, but it seems like I am missing something because the same error is still there. Robert Date: Fri, 18 Jan 2013 09:13:36 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration If you are using the rxfax app then you specify the file name and location: Make sure that the target directory has read/write permissions. You can use something like "chmod 666 /my/directory" to make it read/write. -MC On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: Where does the file get downloaded to? What should the permissions be? Robert > Date: Sat, 19 Jan 2013 00:18:45 +0800 > From: steveu at coppice.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If the TIFF file cannot be opened on receive you probably have a file > permissions issue (e.g. with the directory in which the file is to be > created). If the TIFF file cannot be opened on transmit there are > various possibilities about the suitability of the file's contents. > > Steve > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > This was an incoming fax. I don't have much control over the other > > end. Why is that error appearing when trying to receive something? > > > > Robert > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > wrote: > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > >> Please try pdf, you should be in business now. > >> Please let us know, > >> > >> On Jan 18, 2013 12:35 AM, "Robert Craig" >> > wrote: > >> > >> OK, so I've made a little bit of progress. The distant end is a > >> Windows Fax server hooked up to an analog line. I created a fax > >> extension in FS, assigned an inbound route to the extension, and > >> made sure the extension had an email address. I believe the fax > >> receive wants to complete, but there is an interesting error in > >> the log about TIF files. Anyone ever seen this before? > >> > >> v=0 > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> c=IN IP4 72.201.112.131 > >> t=0 0 > >> m=image 27714 udptl t38 > >> a=T38FaxVersion:0 > >> a=T38MaxBitRate:9600 > >> a=T38FaxFillBitRemoval > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxUdpEC:t38UDPRedundancy > >> > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > >> ip/port confirmed. > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > >> Negotiation Result > >> ======================================================= > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > >> status off > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > >> country: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > >> vendor: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > >> model: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > >> processing not successful - result (41) TIFF/F file cannot be > >> opened.* > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > >> transferred: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > >> fax pages: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > >> resolution: 0x0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > >> status off > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > >> country: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > >> vendor: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > >> model: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > >> sofia/external/+14792460547 @flowroute.com > >> Restore previous codec PCMU:0. > >> EXECUTE sofia/external/+14792460547 > >> @flowroute.com hangup() > >> > >> Robert > >> > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> OK, I am not extremely concerned with getting an ATA working > >> right now. If I could just get an inbound extension-to-email > >> working it would be great. I'll start from scratch and then let > >> you know what happens. Just to make sure, I am going to create an > >> extension, assign an email address to it, make sure there is an > >> inbound route to that extension, and then test it. > >> > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > >> From: gassaad at emassembly.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> The reason for outbound extension is: > >> When you send a fax from ATA to the outside world, it will dial > >> first to the outbound extension and then after successful receive > >> it will dial out to the intended number. > >> > >> As for the error you have it may be related to your ATA. I am not > >> sure please see why do you get the following message. > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > >> > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > >> > wrote: > >> > >> OK, I've already tried creating one extension for inbound and > >> set an email address. Of course, it failed and the errors are > >> at the bottom of this email. What is the purpose of the > >> outbound extension? > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > >> From: gassaad at emassembly.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Sorry for the delay to get back to you. > >> 1- App --> fax server > >> 2- Create two extensions (one for inbound) another for > >> (outbound) From within either extension you can send faxes > >> "virtual fax" > >> 3- point your create your outbound and inbound dialplans as > >> normal but make it to point to those extensions. > >> > >> Good luck, > >> > >> George > >> > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > >> > > >> wrote: > >> > >> What did you do on the config side to get it to work? > >> I've tried both an Ata and a fax to email extension and > >> they both fail in both directions. > >> > >> Robert > >> > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > >> > > >> wrote: > >> > >> I use Fusion successfully to send / receive fax. > >> The way it works: > >> ATA --->FaxServer. > >> After successful receipt, Fax Server---> DID where > >> you have dialled originally. > >> > >> Cheers, > >> > >> George > >> > >> > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > >> >> > wrote: > >> > >> Did anyone see the issue or what I may have > >> missed in the config? As mentioned earlier, I > >> thought it would after a fresh install. Any help > >> is appreciated. > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Thanks for the quick response. Below is the > >> capture from the log viewer. In this test, I am > >> using a remote fax machine (line from Cox with > >> physical machine) faxing into the "email server" > >> where I created an extension for fax-to-email. > >> Calling number is 1479-246-0547. > >> > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:99 > >> sofia/external/+14792460547 at flowroute.com > >> Standard DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> mod_sofia.c:396 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:559 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State Change > >> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:556 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate Change HANGUP > >> -> DOWN > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1510 Close Channel > >> sofia/external/+14792460547 at flowroute.com > >> [CS_DESTROY] > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1506 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Ended > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1488 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Locked, Waiting on > >> external entities > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:440 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change CS_REPORTING > >> -> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State > >> REPORTING going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:92 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> REPORTING, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:446 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_HANGUP -> CS_REPORTING > >> > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:599 Hangup Command > >> with no Session system(/usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 > >> name=+14792460547-2013-01-10-13-39-26 > >> messages='result: Disconnected after permitted > >> retries sender:ATW-4792460547 pages:0' > >> domain=192.168.4.15 caller_id_name='' > >> caller_id_number=): > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:48 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> HANGUP, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:552 Sending BYE to > >> sofia/external/+14792460547 at flowroute.com > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:503 Channel > >> sofia/external/+14792460547 at flowroute.com > >> hanging up, > >> cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:2687 > >> sofia/external/+14792460547 at flowroute.com > >> skip receive > >> message [APPLICATION_EXEC_COMPLETE] (channel is > >> hungup already) > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:3017 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [KILL] > >> 2013-01-10 13:39:52.216151 [NOTICE] > >> mod_dptools.c:1150 Hangup > >> sofia/external/+14792460547 at flowroute.com > >> [CS_EXECUTE] > >> [NORMAL_CLEARING] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:2994 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change ACTIVE -> HANGUP > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> hangup() > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_codec.c:244 > >> sofia/external/+14792460547 at flowroute.com > >> Restore > >> previous codec PCMU:0. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:516 remote model: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:515 remote vendor: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:514 remote country: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:513 ECM status off > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:509 Total fax pages: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:507 Pages transferred: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > >> Fax Ident > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:505 Remote station id: > >> ATW-4792460547 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:500 Fax processing not > >> successful - result (48) Disconnected after > >> permitted retries. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> 2013-01-10 13:39:38.766190 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:29.576127 [ERR] > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > >> must be between 20 and 10000 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxRateManagement:transferredTCF > >> m=image 31632 udptl t38 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completing][200] > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > >> sofia/external/+14792460547 at flowroute.com > >> Update Callee > >> ID to "+14792460547" <+14792460547> > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [calling][0] > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:2647 Local SDP: > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1379 Raw write codec activation > >> Success L16 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_core_codec.c:219 > >> sofia/external/+14792460547 at flowroute.com > >> Push codec L16:70 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1363 Raw read codec activation > >> Success L16 20000 > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > >> > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_ivr_play_say.c:1682 done playing file > >> silence_stream://2000 > >> 2013-01-10 13:39:26.256197 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> switch_ivr_play_say.c:1309 Codec Activated > >> L16 at 8000hz 1 channels 20ms > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> playback(silence_stream://2000) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38_request]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38_request=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38]=[true] > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completed][200] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.006127 [NOTICE] > >> mod_dptools.c:1176 Channel > >> [sofia/external/+14792460547 at flowroute.com > >> ] has been > >> answered > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_channel.c:3395 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change RINGING -> ACTIVE > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=sendrecv > >> a=ptime:20 > >> a=silenceSupp:off - - - - > >> a=fmtp:101 0-16 > >> a=rtpmap:101 telephone-event/8000 > >> a=rtpmap:0 PCMU/8000 > >> m=audio 27588 RTP/AVP 0 101 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_sofia.c:856 Local SDP > >> sofia/external/+14792460547 at flowroute.com > >> : > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:1046 STUN Success > >> [72.201.112.131]:[27588] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> answer() > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [api_hangup_hook]=[system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=${last_fax} messages='result: > >> ${fax_result_text} > >> sender:${fax_remote_station_id} > >> pages:${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='${caller_id_name}' > >> caller_id_number=${caller_id_number}] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\${last_fax} > >> messages='result: \${fax_result_text} > >> sender:\${fax_remote_station_id} > >> pages:\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\${caller_id_name}' > >> caller_id_number=\${caller_id_number}) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [call_direction]=[local] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(call_direction=local) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action hangup() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> playback(silence_stream://2000) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38_request=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action answer() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\\\${last_fax} > >> messages='result: \\\${fax_result_text} > >> sender:\\\${fax_remote_station_id} > >> pages:\\\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\\\${caller_id_name}' > >> caller_id_number=\\\${caller_id_number}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test Fax] destination_number(3150) =~ > >> /^3150$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Test Fax] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain_user] destination_number(3150) =~ > >> /^\*97$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain_user] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain] destination_number(3150) =~ > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [send_to_voicemail] > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->send_to_voicemail] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [extension-intercom] > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->extension-intercom] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [intercept-ext] destination_number(3150) > >> =~ /^\*\*(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->intercept-ext] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_return] destination_number(3150) =~ > >> /^\*69$|^lcr$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_return] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_privacy] destination_number(3150) =~ > >> /^\*67(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_privacy] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [global] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > >> break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->global] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [redial] destination_number(3150) =~ > >> /^(redial|\*870)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->redial] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [group-intercept] destination_number(3150) > >> =~ /^\*8$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->group-intercept] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> ANTI-Action > >> set(call_direction=local) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call-direction] ${call_direction}() =~ > >> /^(inbound|outbound|local)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call-direction] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [Flowroute.11d] destination_number(3150) > >> =~ /^\+?(\d{11})$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Flowroute.11d] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->3150 in context default > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:25.816280 [NOTICE] > >> switch_ivr.c:1781 Transfer > >> sofia/external/+14792460547 at flowroute.com > >> to > >> XML[3150 at default] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_ivr.c:1774 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_EXECUTE -> CS_ROUTING > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> transfer(3150 > >> XML default) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_channel.c:1135 EXPORT (export_vars) > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [outside_call]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(outside_call=true) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> transfer(3150 XML default) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] > >> destination_number(15203039810) =~ > >> /^15203039810$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] context(public) =~ /public/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->Test_Number] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_debug] ${call_debug}(false) =~ > >> /^true$/ break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->call_debug] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(outside_call=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [outside_call] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->outside_call] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->unloop] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->15203039810 in context public > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_channel.c:2003 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change DOWN -> RINGING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:454 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State INIT > >> going to sleep > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> mod_sofia.c:126 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_INIT -> CS_ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:454 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_NEW -> CS_INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5263 Set 2833 dtmf send/recv payload > >> to 101 > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_codec.c:111 > >> sofia/external/+14792460547 at flowroute.com > >> Original read > >> codec set to PCMU:0 > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:3093 Set Codec > >> sofia/external/+14792460547 at flowroute.com > >> PCMU/8000 20 > >> ms 160 samples 64000 bits > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5134 Audio Codec Compare > >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5134 Audio Codec Compare > >> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > >> > >> a=ptime:20 > >> a=fmtp:101 0-15 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:18 annexb=no > >> a=rtpmap:18 G729/8000 > >> m=audio 31632 RTP/AVP 0 18 101 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849875 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [received][100] > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_state_machine.c:433 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State NEW > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_NEW > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.786370 [NOTICE] > >> switch_channel.c:968 New Channel > >> sofia/external/+14792460547 at flowroute.com > >> > >> [d276f012-5b65-11e2-beae-8564ff827a25] > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Thu, 10 Jan 2013 09:47:18 -0800 > >> From: msc at freeswitch.org > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Can you pastebin a console debug log w/ sip trace > >> of a call failing to negotiate t.38 correctly? > >> That might help us figure out where to look. > >> -MC > >> > >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig > >> >> > wrote: > >> > >> Just installed FusionPBX using the script > >> built for a Centos 6 install. Everything > >> works great in regards to calls, but my main > >> goal of this project was to utilize the fax > >> capabilities of Freeswitch/FusionPBX. I have > >> tried many things in media bypass, routing to > >> a fax extension, using an ATA, etc., but I > >> can't get T38 to negotiate correctly. Can > >> someone point me in the right direction as to > >> where I should begin? I am guessing it isn't > >> supposed to work right out of the box. I am > >> using Flowroute as my provider right now and > >> the have confirmed that they utilize T38 in > >> both directions. Thanks! > >> > >> > >> Robert > >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/ce3fa9f3/attachment-0001.html From bdfoster at endigotech.com Fri Jan 18 20:59:38 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 18 Jan 2013 12:59:38 -0500 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: <50F975E5.2040206@coppice.org> Message-ID: <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com> Submit the dialplan you are using to receive the fax. You can use the FreeSWITCH pastebin, http://pastebin.freeswitch.org Sent from my iPhone On Jan 18, 2013, at 12:43 PM, Robert Craig wrote: > I believe I'm just using the built in SpanDSP for faxing. I've looked around for 'rxfax' and can't find anything. I did notice that the directory where the faxes were supposed to be stored for the extension I create didn't even exist. I created them, applied the permissions, but it seems like I am missing something because the same error is still there. > > > Robert > > > > Date: Fri, 18 Jan 2013 09:13:36 -0800 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If you are using the rxfax app then you specify the file name and location: > > > > Make sure that the target directory has read/write permissions. You can use something like "chmod 666 /my/directory" to make it read/write. > > -MC > > On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: > Where does the file get downloaded to? What should the permissions be? > > Robert > > > > Date: Sat, 19 Jan 2013 00:18:45 +0800 > > From: steveu at coppice.org > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] T38 Configuration > > > > If the TIFF file cannot be opened on receive you probably have a file > > permissions issue (e.g. with the directory in which the file is to be > > created). If the TIFF file cannot be opened on transmit there are > > various possibilities about the suitability of the file's contents. > > > > Steve > > > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > > This was an incoming fax. I don't have much control over the other > > > end. Why is that error appearing when trying to receive something? > > > > > > Robert > > > > > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > > wrote: > > > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > > >> Please try pdf, you should be in business now. > > >> Please let us know, > > >> > > >> On Jan 18, 2013 12:35 AM, "Robert Craig" > >> > wrote: > > >> > > >> OK, so I've made a little bit of progress. The distant end is a > > >> Windows Fax server hooked up to an analog line. I created a fax > > >> extension in FS, assigned an inbound route to the extension, and > > >> made sure the extension had an email address. I believe the fax > > >> receive wants to complete, but there is an interesting error in > > >> the log about TIF files. Anyone ever seen this before? > > >> > > >> v=0 > > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> c=IN IP4 72.201.112.131 > > >> t=0 0 > > >> m=image 27714 udptl t38 > > >> a=T38FaxVersion:0 > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxFillBitRemoval > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > > >> ip/port confirmed. > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > > >> Negotiation Result > > >> ======================================================= > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > > >> status off > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > > >> country: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > > >> vendor: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > > >> model: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > > >> ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > > >> ============================================================================== > > >> > > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > > >> processing not successful - result (41) TIFF/F file cannot be > > >> opened.* > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > > >> transferred: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > > >> fax pages: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > > >> resolution: 0x0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > > >> status off > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > > >> country: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > > >> vendor: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > > >> model: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > > >> ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > > >> sofia/external/+14792460547 @flowroute.com > > >> Restore previous codec PCMU:0. > > >> EXECUTE sofia/external/+14792460547 > > >> @flowroute.com hangup() > > >> > > >> Robert > > >> > > >> > > >> > > >> ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> OK, I am not extremely concerned with getting an ATA working > > >> right now. If I could just get an inbound extension-to-email > > >> working it would be great. I'll start from scratch and then let > > >> you know what happens. Just to make sure, I am going to create an > > >> extension, assign an email address to it, make sure there is an > > >> inbound route to that extension, and then test it. > > >> > > >> > > >> Robert > > >> > > >> > > >> ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > > >> From: gassaad at emassembly.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> The reason for outbound extension is: > > >> When you send a fax from ATA to the outside world, it will dial > > >> first to the outbound extension and then after successful receive > > >> it will dial out to the intended number. > > >> > > >> As for the error you have it may be related to your ATA. I am not > > >> sure please see why do you get the following message. > > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > > >> > > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > > >> > wrote: > > >> > > >> OK, I've already tried creating one extension for inbound and > > >> set an email address. Of course, it failed and the errors are > > >> at the bottom of this email. What is the purpose of the > > >> outbound extension? > > >> > > >> Robert > > >> > > >> > > >> ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > > >> From: gassaad at emassembly.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Sorry for the delay to get back to you. > > >> 1- App --> fax server > > >> 2- Create two extensions (one for inbound) another for > > >> (outbound) From within either extension you can send faxes > > >> "virtual fax" > > >> 3- point your create your outbound and inbound dialplans as > > >> normal but make it to point to those extensions. > > >> > > >> Good luck, > > >> > > >> George > > >> > > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > > >> > > > >> wrote: > > >> > > >> What did you do on the config side to get it to work? > > >> I've tried both an Ata and a fax to email extension and > > >> they both fail in both directions. > > >> > > >> Robert > > >> > > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > > >> > > > >> wrote: > > >> > > >> I use Fusion successfully to send / receive fax. > > >> The way it works: > > >> ATA --->FaxServer. > > >> After successful receipt, Fax Server---> DID where > > >> you have dialled originally. > > >> > > >> Cheers, > > >> > > >> George > > >> > > >> > > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > > >> > >> > wrote: > > >> > > >> Did anyone see the issue or what I may have > > >> missed in the config? As mentioned earlier, I > > >> thought it would after a fresh install. Any help > > >> is appreciated. > > >> > > >> Robert > > >> > > >> > > >> ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Thanks for the quick response. Below is the > > >> capture from the log viewer. In this test, I am > > >> using a remote fax machine (line from Cox with > > >> physical machine) faxing into the "email server" > > >> where I created an extension for fax-to-email. > > >> Calling number is 1479-246-0547. > > >> > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:99 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> mod_sofia.c:396 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:559 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State Change > > >> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:556 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate Change HANGUP > > >> -> DOWN > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1510 Close Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_DESTROY] > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1506 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Ended > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1488 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Locked, Waiting on > > >> external entities > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:440 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change CS_REPORTING > > >> -> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State > > >> REPORTING going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:92 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> REPORTING, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:446 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_HANGUP -> CS_REPORTING > > >> > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:599 Hangup Command > > >> with no Session system(/usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 > > >> name=+14792460547-2013-01-10-13-39-26 > > >> messages='result: Disconnected after permitted > > >> retries sender:ATW-4792460547 pages:0' > > >> domain=192.168.4.15 caller_id_name='' > > >> caller_id_number=): > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:48 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> HANGUP, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:552 Sending BYE to > > >> sofia/external/+14792460547 at flowroute.com > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:503 Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> hanging up, > > >> cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:2687 > > >> sofia/external/+14792460547 at flowroute.com > > >> skip receive > > >> message [APPLICATION_EXEC_COMPLETE] (channel is > > >> hungup already) > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:3017 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [KILL] > > >> 2013-01-10 13:39:52.216151 [NOTICE] > > >> mod_dptools.c:1150 Hangup > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_EXECUTE] > > >> [NORMAL_CLEARING] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:2994 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change ACTIVE -> HANGUP > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> hangup() > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_codec.c:244 > > >> sofia/external/+14792460547 at flowroute.com > > >> Restore > > >> previous codec PCMU:0. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:518 > > >> ============================================================================== > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:516 remote model: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:515 remote vendor: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:514 remote country: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:513 ECM status off > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:509 Total fax pages: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:507 Pages transferred: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > > >> Fax Ident > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:505 Remote station id: > > >> ATW-4792460547 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:500 Fax processing not > > >> successful - result (48) Disconnected after > > >> permitted retries. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:487 > > >> ============================================================================== > > >> > > >> 2013-01-10 13:39:38.766190 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:29.576127 [ERR] > > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > > >> must be between 20 and 10000 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxRateManagement:transferredTCF > > >> m=image 31632 udptl t38 > > >> t=0 0 > > >> c=IN IP4 66.42.121.5 > > >> s=- > > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > > >> v=0 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > > >> Remote SDP: > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completing][200] > > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > > >> sofia/external/+14792460547 at flowroute.com > > >> Update Callee > > >> ID to "+14792460547" <+14792460547> > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [calling][0] > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:2647 Local SDP: > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1379 Raw write codec activation > > >> Success L16 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_core_codec.c:219 > > >> sofia/external/+14792460547 at flowroute.com > > >> Push codec L16:70 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1363 Raw read codec activation > > >> Success L16 20000 > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > > >> > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_ivr_play_say.c:1682 done playing file > > >> silence_stream://2000 > > >> 2013-01-10 13:39:26.256197 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> switch_ivr_play_say.c:1309 Codec Activated > > >> L16 at 8000hz 1 channels 20ms > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> playback(silence_stream://2000) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38_request]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38_request=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38]=[true] > > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completed][200] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.006127 [NOTICE] > > >> mod_dptools.c:1176 Channel > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] has been > > >> answered > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_channel.c:3395 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change RINGING -> ACTIVE > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=sendrecv > > >> a=ptime:20 > > >> a=silenceSupp:off - - - - > > >> a=fmtp:101 0-16 > > >> a=rtpmap:101 telephone-event/8000 > > >> a=rtpmap:0 PCMU/8000 > > >> m=audio 27588 RTP/AVP 0 101 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_sofia.c:856 Local SDP > > >> sofia/external/+14792460547 at flowroute.com > > >> : > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:1046 STUN Success > > >> [72.201.112.131]:[27588] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> answer() > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [api_hangup_hook]=[system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=${last_fax} messages='result: > > >> ${fax_result_text} > > >> sender:${fax_remote_station_id} > > >> pages:${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='${caller_id_name}' > > >> caller_id_number=${caller_id_number}] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\${last_fax} > > >> messages='result: \${fax_result_text} > > >> sender:\${fax_remote_station_id} > > >> pages:\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\${caller_id_name}' > > >> caller_id_number=\${caller_id_number}) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [call_direction]=[local] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(call_direction=local) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action hangup() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> playback(silence_stream://2000) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38_request=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action answer() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\\\${last_fax} > > >> messages='result: \\\${fax_result_text} > > >> sender:\\\${fax_remote_station_id} > > >> pages:\\\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\\\${caller_id_name}' > > >> caller_id_number=\\\${caller_id_number}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test Fax] destination_number(3150) =~ > > >> /^3150$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Test Fax] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain_user] destination_number(3150) =~ > > >> /^\*97$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain_user] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain] destination_number(3150) =~ > > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [send_to_voicemail] > > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->send_to_voicemail] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [extension-intercom] > > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->extension-intercom] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [intercept-ext] destination_number(3150) > > >> =~ /^\*\*(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->intercept-ext] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_return] destination_number(3150) =~ > > >> /^\*69$|^lcr$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_return] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_privacy] destination_number(3150) =~ > > >> /^\*67(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_privacy] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [global] > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > > >> break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->global] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [redial] destination_number(3150) =~ > > >> /^(redial|\*870)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->redial] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [group-intercept] destination_number(3150) > > >> =~ /^\*8$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->group-intercept] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> ANTI-Action > > >> set(call_direction=local) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call-direction] ${call_direction}() =~ > > >> /^(inbound|outbound|local)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call-direction] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [Flowroute.11d] destination_number(3150) > > >> =~ /^\+?(\d{11})$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Flowroute.11d] continue=false > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->3150 in context default > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [NOTICE] > > >> switch_ivr.c:1781 Transfer > > >> sofia/external/+14792460547 at flowroute.com > > >> to > > >> XML[3150 at default] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_ivr.c:1774 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_EXECUTE -> CS_ROUTING > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> transfer(3150 > > >> XML default) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_channel.c:1135 EXPORT (export_vars) > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [outside_call]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(outside_call=true) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> transfer(3150 XML default) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] > > >> destination_number(15203039810) =~ > > >> /^15203039810$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] context(public) =~ /public/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->Test_Number] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_debug] ${call_debug}(false) =~ > > >> /^true$/ break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->call_debug] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(outside_call=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [outside_call] > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->outside_call] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->unloop] continue=false > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->15203039810 in context public > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_channel.c:2003 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change DOWN -> RINGING > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_ROUTING > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:454 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State INIT > > >> going to sleep > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> mod_sofia.c:126 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_INIT -> CS_ROUTING > > >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:454 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_NEW -> CS_INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:5263 Set 2833 dtmf send/recv payload > > >> to 101 > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_codec.c:111 > > >> sofia/external/+14792460547 at flowroute.com > > >> Original read > > >> codec set to PCMU:0 > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:3093 Set Codec > > >> sofia/external/+14792460547 at flowroute.com > > >> PCMU/8000 20 > > >> ms 160 samples 64000 bits > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:5134 Audio Codec Compare > > >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:5134 Audio Codec Compare > > >> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > > >> > > >> a=ptime:20 > > >> a=fmtp:101 0-15 > > >> a=rtpmap:101 telephone-event/8000 > > >> a=fmtp:18 annexb=no > > >> a=rtpmap:18 G729/8000 > > >> m=audio 31632 RTP/AVP 0 18 101 > > >> t=0 0 > > >> c=IN IP4 66.42.121.5 > > >> s=- > > >> o=- 275426305 1357849875 IN IP4 66.42.121.5 > > >> v=0 > > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 > > >> Remote SDP: > > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [received][100] > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_state_machine.c:433 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State NEW > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_NEW > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.786370 [NOTICE] > > >> switch_channel.c:968 New Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> > > >> [d276f012-5b65-11e2-beae-8564ff827a25] > > >> > > >> Robert > > >> > > >> > > >> ------------------------------------------------------------------------ > > >> Date: Thu, 10 Jan 2013 09:47:18 -0800 > > >> From: msc at freeswitch.org > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Can you pastebin a console debug log w/ sip trace > > >> of a call failing to negotiate t.38 correctly? > > >> That might help us figure out where to look. > > >> -MC > > >> > > >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig > > >> > >> > wrote: > > >> > > >> Just installed FusionPBX using the script > > >> built for a Centos 6 install. Everything > > >> works great in regards to calls, but my main > > >> goal of this project was to utilize the fax > > >> capabilities of Freeswitch/FusionPBX. I have > > >> tried many things in media bypass, routing to > > >> a fax extension, using an ATA, etc., but I > > >> can't get T38 to negotiate correctly. Can > > >> someone point me in the right direction as to > > >> where I should begin? I am guessing it isn't > > >> supposed to work right out of the box. I am > > >> using Flowroute as my provider right now and > > >> the have confirmed that they utilize T38 in > > >> both directions. Thanks! > > >> > > >> > > >> Robert > > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/78058336/attachment-0001.html From royj at yandex.ru Fri Jan 18 21:13:40 2013 From: royj at yandex.ru (royj) Date: Fri, 18 Jan 2013 22:13:40 +0400 Subject: [Freeswitch-users] FS hangs up after 30 seconds by timeout on ACK (after sending 200 OK) In-Reply-To: References: Message-ID: <20130118221340.df3c9268e3b47c63ee8b95bd@yandex.ru> Although seems like nat issue (I think), would be interesting to see log with 'fsctl loglevel debug', 'sofia global siptrace on' On Fri, 18 Jan 2013 17:38:55 +0400 ??????? ???????? wrote: > Hi, all! > > I have two sip profiles: profile_spb and profile_msk for one gateway. I > assign sip port 5060 for profile_spb and port 5070 for profile_msk. > > FS has successfully registered two users with distinct usernames: num_spb > and num_msk on the gateway. By the way, field "Contact" in message REGISTER > has normal port number - 5060 and 5070 accordingly. > > When someone calls to num_spb, it shoot to port 5060 and all works well. > When someone calls to num_msk, it shoot to port 5070. Call is received, but > is rejected after 30 seconds by timeout (not received ACK on 200 OK). I > think it's because FS sends 200 OK with port number = 0 in field Contact. > All another headers are good (it seems). > > Does someone know how to say FS to use correct port number in field Contact > in 200 OK? > > Does someone has another version? > > Thanks a lot! -- royj From vbvbrj at gmail.com Fri Jan 18 22:21:36 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 18 Jan 2013 21:21:36 +0200 Subject: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 In-Reply-To: References: <50F909AD.2010308@gmail.com> <50F91490.7080006@gmail.com> Message-ID: <50F9A0C0.8050401@gmail.com> On 18.01.2013 18:19, Steven Ayre wrote: > First of all update to the latest version of v1.2.stable or master - it > may be something that's already fixed if you're on an old version. > > Second, configure FreeSWITCH to save a coredump on a segfault (start it > from the init.d script with the -core param). > > Once you have a coredump you can open a Jira ticket with a backtrace: > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 > The backtrace will show exactly where and why the segfault occurred, > very helpful for debugging. Yes, I updated to latest master. Will have to wait as it does not happen regularly or with some pattern. "-core" option I enabled long ago, but as I said, core files are not writted anywhere. A good option or parameter for FS is to specify where to save core dump files. -- Mimiko desu. From magadum_ashok at yahoo.com Fri Jan 18 23:42:42 2013 From: magadum_ashok at yahoo.com (Ashok M) Date: Fri, 18 Jan 2013 20:42:42 +0000 (UTC) Subject: [Freeswitch-users] =?utf-8?q?trouble_with_record=5Fsession?= Message-ID: I am trying to test recording feature using record_session app. Here is my dial plan for the internal extension that I am trying dial from another extension which gets registered with FreeSWITCH after starting x-lite. The record_session seems to creating the wav file, but if I try to play it back and I don't hear anything. am I doing this right? or am I missing something here? Thanks, -Ashok From vbvbrj at gmail.com Sat Jan 19 00:03:56 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 18 Jan 2013 23:03:56 +0200 Subject: [Freeswitch-users] trouble with record_session In-Reply-To: References: Message-ID: <50F9B8BC.9020403@gmail.com> On 18.01.2013 22:42, Ashok M wrote: > I am trying to test recording feature using record_session app. Here is my > dial plan for the internal extension that I am trying dial from another > extension which gets registered with FreeSWITCH after starting x-lite. The > record_session seems to creating the wav file, but if I try to play it back > and I don't hear anything. am I doing this right? or am I missing something > here? > > > > > > data="d:/record/Site/recordtest.wav"/> > > > > > > > File d:/record/Site/recordtest.wav will be looked for writting until session ends, ie hangup and dialplan execute ends. -- Mimiko desu. From krice at freeswitch.org Sat Jan 19 00:17:15 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 18 Jan 2013 15:17:15 -0600 Subject: [Freeswitch-users] Friday Free For All is Go Message-ID: Get in here... SIP:888 at conference.freeswitch.org Or the normal dialup methods -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/1d1ee302/attachment.html From vbvbrj at gmail.com Sat Jan 19 00:24:25 2013 From: vbvbrj at gmail.com (Mimiko) Date: Fri, 18 Jan 2013 23:24:25 +0200 Subject: [Freeswitch-users] Friday Free For All is Go In-Reply-To: References: Message-ID: <50F9BD89.7010108@gmail.com> On 18.01.2013 23:17, Ken Rice wrote: > Get in here... > > SIP:888 at conference.freeswitch.org > > Or the normal dialup methods Isn't it will magnetic to write: tel:888 at conference.freeswitch.org ? -- Mimiko desu. From freeswitch at orresta.no-ip.com Sat Jan 19 01:00:11 2013 From: freeswitch at orresta.no-ip.com (Jakob) Date: Fri, 18 Jan 2013 23:00:11 +0100 Subject: [Freeswitch-users] levels of ? Message-ID: <50F9C5EB.5020902@orresta.no-ip.com> Hi, Can you make a construct like below? Match the second argument within a match statement. Or are there a switch statement that can be used? Best Regards Jakob From steveayre at gmail.com Sat Jan 19 01:10:20 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 18 Jan 2013 22:10:20 +0000 Subject: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 In-Reply-To: <50F9A0C0.8050401@gmail.com> References: <50F909AD.2010308@gmail.com> <50F91490.7080006@gmail.com> <50F9A0C0.8050401@gmail.com> Message-ID: <1C7BD7EC-8865-40EE-A94D-5D2D785B99E4@gmail.com> They should go to the cwd, get the FS process and find it by 'ls -ld /proc/PID/cwd' Also check the ulimit settings, -c limits the size of coredumps. Steve on iPhone On 18 Jan 2013, at 19:21, Mimiko wrote: > On 18.01.2013 18:19, Steven Ayre wrote: >> First of all update to the latest version of v1.2.stable or master - it >> may be something that's already fixed if you're on an old version. >> >> Second, configure FreeSWITCH to save a coredump on a segfault (start it >> from the init.d script with the -core param). >> >> Once you have a coredump you can open a Jira ticket with a backtrace: >> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 >> The backtrace will show exactly where and why the segfault occurred, >> very helpful for debugging. > > Yes, I updated to latest master. Will have to wait as it does not happen > regularly or with some pattern. "-core" option I enabled long ago, but > as I said, core files are not writted anywhere. A good option or > parameter for FS is to specify where to save core dump files. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lists at kavun.ch Sat Jan 19 01:18:13 2013 From: lists at kavun.ch (Emrah) Date: Fri, 18 Jan 2013 17:18:13 -0500 Subject: [Freeswitch-users] Cannot save Voicemail message Message-ID: <3263505F-2AC8-4BB5-ABEB-44B706AD80C5@kavun.ch> Hi all, I've noticed a strange problem lately, I can't press 3 to save a VM. I use mod_shout, my VMs are in MP3 format. Log abstract: 2013-01-18 17:11:09.191700 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 16000hz 1 channels 20ms 2013-01-18 17:11:10.191704 [DEBUG] switch_rtp.c:3827 RTP RECV DTMF 3:1120 2013-01-18 17:11:10.191704 [DEBUG] switch_ivr_play_say.c:1682 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-save_recording.wav 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-you_have.wav] (en:en) freeswitch at internal> version FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z) Deleting works fine. Any idea? Thanks and all the best, Emrah From cal.leeming at simplicitymedialtd.co.uk Sat Jan 19 01:18:55 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 18 Jan 2013 22:18:55 +0000 Subject: [Freeswitch-users] mod_xml_cdr and transfers - missing call legs Message-ID: Hello, So, we are currently testing mod_xml_cdr and having some difficulties getting all the correct data. I actually have a fairly well commented dump of this happening (albeit from an unrelated bug); http://jira.freeswitch.org/secure/attachment/17781/debug-18-01-2013.txt Take the following scenario: >From user/2000 TO user/2001 from user/2001 to user/2002 via user/2000 from user/2002 to gw/447534971120 via user/2001 from gw/447534971120 to ivr/2003 via user/2001 from gw/447534971120 to gw/442477062086 via ivr/2003 HANGUP from gw/447534971120 At this point the CDRs are posted back to our web app (both A and B leg).. however, it doesn't seem to contain all the above call events.. The end goal for me was to have each stage of a call logged and linked.. so if a call is transferred 10 times, I will see 10 CDR legs. So, my questions are; * Is this behaviour expected? * Am I approaching this from the wrong angle? * Are there better ways to collect CDRs? Any and all feedback would be much appreciated. Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/ee78f138/attachment-0001.html From msc at freeswitch.org Sat Jan 19 01:47:41 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Jan 2013 14:47:41 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130116025242.83fde4e4@mail.tritonwest.net> Message-ID: You need an ATA that hopefully won't mangle your fax transmissions. I've heard that the SPA 112's are pretty good, but that's only anecdotal. At least they are inexpensive, so it's not a large investment to give it a try. -MC On Wed, Jan 16, 2013 at 3:24 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Tue, Jan 15, 2013 at 6:52 PM, Dave R. Kompel wrote: > >> ** >> Another option would be on the Linux box, enable SPANDSP modems, and use >> com0com on windows to backup com ports to raw TCP connections. It supports >> both. That way it would talk over a TCP pipe to an emulated modem on your >> other FreeSWITCH box. >> > > That sounds like the way to go. I'll give that a try. > > But first, I have another FAX problem. The users here use a real FAX > machine to send FAXes. Since we're getting rid of the POTS lines, I > attempted to use a (cheap) OBi100 to connect the analog FAX machine to the > FreeSwitch box. It didn't work. Voice calls work, but it can't seem to > sync with the remote FAX machine. Any ideas how to get a regular FAX > machine to work with VoIP? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/56bb3da4/attachment.html From msc at freeswitch.org Sat Jan 19 01:56:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Jan 2013 14:56:37 -0800 Subject: [Freeswitch-users] Unloop extension In-Reply-To: <20130117100537.56256009@mail.btcom.kz> References: <20130117100537.56256009@mail.btcom.kz> Message-ID: Yes, that needs to be rephrased. It really should be ${var} not $${var} since $${var} is meant to address a different scenario. -MC On Wed, Jan 16, 2013 at 8:05 PM, ????? ??????? wrote: > FreeSWITCH 1.0.6 book on p. 187 says: > > ... When you set a variable during this phase, that variable is considered > global automatically and becomes accessible throughout the application as > $${variable} elsewhere in the XML. > > ... You can utilize global variables in your conditions, your variable and > parameter declarations, pretty much anywhere, like this: > > > This should had an addition, that ${g_v} can be used as well. > > ----- Original Message ----- > From: Michael Collins [mailto:msc at freeswitch.org] > To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > Sent: Thu, 17 Jan 2013 07:17:31 +0600 > Subject: Re: [Freeswitch-users] Unloop extension > > > > No, it's available globally, whether used with $${} or ${} notation. You > > can also see it from fs_cli: > > > > freeswitch at default> global_getvar unroll_loops > > true > > > > More info about $${} vs. ${} is found > > here< > http://wiki.freeswitch.org/wiki/Channel_Variables#.24.7Bvariable.7D_vs._.24.24.7Bvariable.7D > > > > . > > > > -MC > > > > On Tue, Jan 15, 2013 at 8:09 PM, ????? ??????? > > wrote: > > > > > Ain't what was set with when > > > used in $${...} notation? > > > > > > ----- Original Message ----- > > > From: Michael Collins [mailto:msc at freeswitch.org] > > > To: FreeSWITCH Users Help [mailto: > freeswitch-users at lists.freeswitch.org] > > > Sent: Wed, 16 Jan 2013 06:54:12 +0600 > > > Subject: Re: [Freeswitch-users] Unloop extension > > > > > > > > > > Yes, it's okay. The pre-process does a set and ${unroll_loops} gets > set > > > as > > > > needed. Then this dp entry does the work: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > More info on sip_looped_call var: > > > > http://wiki.freeswitch.org/wiki/Variable_sip_looped_call > > > > > > > > -MC > > > > > > > > On Mon, Jan 14, 2013 at 9:33 PM, ????? ??????? > > > > wrote: > > > > > > > > > What unloop extension is used for? BTW: vars.xml pre-(process)-sets > > > > > unroll_loops, whereas DPs (public and default) are using channel > > > variable > > > > > syntax? Is this ok? > > > > > > > > > > Y. > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > Professional FreeSWITCH Consulting Services: > > > > > consulting at freeswitch.org > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > http://www.freeswitch.org > > > > > http://wiki.freeswitch.org > > > > > http://www.cluecon.com > > > > > > > > > > FreeSWITCH-users mailing list > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > -- > > > > Michael S Collins > > > > Twitter: @mercutioviz > > > > http://www.FreeSWITCH.org > > > > http://www.ClueCon.com > > > > http://www.OSTAG.org > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/d8d9d65b/attachment.html From msc at freeswitch.org Sat Jan 19 02:01:36 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Jan 2013 15:01:36 -0800 Subject: [Freeswitch-users] SIP Message Parsing and Auto-Action In-Reply-To: References: Message-ID: This smells like a job for an ESL-based daemon. The daemon could send OPTIONs pings to the HT and then issue the reboot command any time it detects that warning header. -MC On Thu, Jan 17, 2013 at 7:08 AM, Nick Vines wrote: > I have a couple Grandstream HT502s that occasionally get stuck with the > detected NAT type of > Warning: 399 ip.address "Detected NAT type is UDP Blocked". When that > happens the device is never able to complete a call until it is restarted. > Where should I start looking to create a monitor that will parse those > messages and send a reboot (i.e. sofia profile [profile] flush_inbound_reg > [call_id] reboot) > > Thanks, > Nick > > *Sample SIP Message* > SIP/2.0 200 OK > Via: SIP/2.0/TCP XXX.XXX.XXX.XXX:PORT;branch=z9hG4bK2N9vy7BmyF40B > From: ;tag=3yNZa6ZNv6K6g > To: ;tag=1662499825 > Call-ID: 3901be5e-db58-1230-109f-bc764e103a41 > CSeq: 38881081 NOTIFY > Supported: replaces, path, timer, eventlist > User-Agent: Grandstream HT-502 V1.2A 1.0.8.4 chip V2.2 > Warning: 399 XXX.XXX.XXX.XXX *"Detected NAT type is UDP Blocked"* > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > REFER, UPDATE > Content-Length: 0 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/85824c2e/attachment-0001.html From msc at freeswitch.org Sat Jan 19 02:10:46 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Jan 2013 15:10:46 -0800 Subject: [Freeswitch-users] mod_xml_curl - request for comments In-Reply-To: References: <804D48104511D4468F0D60DF9D3100350B0A1102@MAIL.millicorp.com> Message-ID: Anything that is globally applicable probably shouldn't be documented solely on this page. It would be better to codify that in a separate page that can be linked-to from various other pages. I would cut & paste that information into a diff wiki page and link to it from the xml curl page and see what that looks like. -MC On Fri, Jan 18, 2013 at 9:34 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > I agree that having some alternative input/ideas on this page would > certainly be a good thing. > > I'm almost tempted to say that this page doesn't need to explain/detail > any of the channel variable contexts (the ones where it explains Applicable > common fields etc), as this is really FreeSWITCH wide, > and interpretation of these variables mostly depends on your own business > logic. > > Removing this info would make the page a lot more read-able, but at the > same time, might make it more difficult to understand for new users. > > Perhaps a warning at the top somewhere explaining this might be more > suitable? > > Cal > > On Fri, Jan 18, 2013 at 4:49 PM, Michael Collins wrote: > >> Agreed, this is a significant improvement. Keep plugging away and if >> anyone else has the time and knowledge to add to this page that would be >> fantastic. >> -MC >> >> On Fri, Jan 18, 2013 at 6:50 AM, Tim Meade wrote: >> >>> Hi Cal?.**** >>> >>> ** ** >>> >>> I think it looks MUCH better. **** >>> >>> ** ** >>> >>> Great job,**** >>> >>> ** ** >>> >>> Tim**** >>> >>> ** ** >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal >>> Leeming [Simplicity Media Ltd] >>> *Sent:* Friday, January 18, 2013 9:33 AM >>> *To:* FreeSWITCH Users Help >>> *Subject:* [Freeswitch-users] mod_xml_curl - request for comments**** >>> >>> ** ** >>> >>> Hello,**** >>> >>> ** ** >>> >>> I've been re-reading my changes to mod_xml_curl documentation, and I'm >>> still not entirely happy with it.**** >>> >>> ** ** >>> >>> I can't tell if it genuinely needs a different approach, or if I'm just >>> being fussy.**** >>> >>> ** ** >>> >>> Therefore, constructive criticism is very much welcome - even if it >>> means the whole damn thing has to be re-done again.**** >>> >>> ** ** >>> >>> http://wiki.freeswitch.org/wiki/Mod_xml_curl**** >>> >>> ** ** >>> >>> Thanks**** >>> >>> >>> Cal**** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/5c205544/attachment.html From lconroy at insensate.co.uk Sat Jan 19 02:13:30 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Fri, 18 Jan 2013 23:13:30 +0000 Subject: [Freeswitch-users] Friday Free For All is Go In-Reply-To: <50F9BD89.7010108@gmail.com> References: <50F9BD89.7010108@gmail.com> Message-ID: <5B4B9DBA-D2C8-446A-B42D-EB27B009B072@insensate.co.uk> On 18 Jan 2013, at 21:24, Mimiko wrote: > Isn't it will magnetic to write: tel:888 at conference.freeswitch.org ? To which I reply: I have no idea what "will magnetic" means, but to be syntactically correct (see ), that URL would need to be: I haven't checked to see if fS handles a phone-context parameter of this form, but no-one's perfect :). all the best, Lawrence From lconroy at insensate.co.uk Sat Jan 19 02:16:13 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Fri, 18 Jan 2013 23:16:13 +0000 Subject: [Freeswitch-users] Friday Free For All is Go In-Reply-To: <50F9BD89.7010108@gmail.com> References: <50F9BD89.7010108@gmail.com> Message-ID: <7242B074-25D6-472D-A978-3811607550D9@insensate.co.uk> On 18 Jan 2013, at 21:24, Mimiko wrote: > Isn't it will magnetic to write: tel:888 at conference.freeswitch.org ? To which I reply: -- take 2, without typo I have no idea what "will magnetic" means, but to be syntactically correct (see ), that URL would need to be: I haven't checked to see if fS handles a phone-context parameter of this form, but no-one's perfect :). all the best, Lawrence From msc at freeswitch.org Sat Jan 19 02:15:45 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Jan 2013 15:15:45 -0800 Subject: [Freeswitch-users] levels of ? In-Reply-To: <50F9C5EB.5020902@orresta.no-ip.com> References: <50F9C5EB.5020902@orresta.no-ip.com> Message-ID: No, you can't nest them like that. But you can stack multiple patterns inside a pattern and use break_on_match="true" to prevent further matching attempts inside that macro. Look at "voicemail_message_count" macro in the vanilla configs for a good example of that. -MC On Fri, Jan 18, 2013 at 2:00 PM, Jakob wrote: > Hi, > > Can you make a construct like below? Match the second argument within a > match statement. Or are there a switch statement that can be used? > > > > > > > > > > > > > > > > > > > > > > > Best Regards > Jakob > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/1aac329f/attachment-0001.html From msc at freeswitch.org Sat Jan 19 02:24:15 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Jan 2013 15:24:15 -0800 Subject: [Freeswitch-users] Cannot save Voicemail message In-Reply-To: <3263505F-2AC8-4BB5-ABEB-44B706AD80C5@kavun.ch> References: <3263505F-2AC8-4BB5-ABEB-44B706AD80C5@kavun.ch> Message-ID: Can you supply a bit more context? Show us more log lines. -MC On Fri, Jan 18, 2013 at 2:18 PM, Emrah wrote: > Hi all, > > I've noticed a strange problem lately, I can't press 3 to save a VM. > > I use mod_shout, my VMs are in MP3 format. > > Log abstract: > > 2013-01-18 17:11:09.191700 [DEBUG] switch_ivr_play_say.c:1309 Codec > Activated L16 at 16000hz 1 channels 20ms > 2013-01-18 17:11:10.191704 [DEBUG] switch_rtp.c:3827 RTP RECV DTMF 3:1120 > 2013-01-18 17:11:10.191704 [DEBUG] switch_ivr_play_say.c:1682 done playing > file > /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-save_recording.wav > 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:67 No language > specified - Using [en] > 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[voicemail/vm-you_have.wav] (en:en) > > freeswitch at internal> version > FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 > 2012-12-29 00:17:59Z) > > Deleting works fine. > Any idea? > > Thanks and all the best, > Emrah > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/3a320811/attachment.html From bdfoster at endigotech.com Sat Jan 19 02:42:22 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 18 Jan 2013 18:42:22 -0500 Subject: [Freeswitch-users] Friday Free For All is Go In-Reply-To: References: Message-ID: Conference needs a dial up DID to remind us of the AOL days!!! Sent from my iPhone On Jan 18, 2013, at 4:17 PM, Ken Rice wrote: > Get in here... > > SIP:888 at conference.freeswitch.org > > Or the normal dialup methods > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/c1366c2d/attachment.html From krice at freeswitch.org Sat Jan 19 03:15:44 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 18 Jan 2013 18:15:44 -0600 Subject: [Freeswitch-users] Friday Free For All is Go In-Reply-To: Message-ID: The conference has several DIDs on it... I think ?+1.919.386.9900 is one of them On 1/18/13 5:42 PM, "Brian Foster" wrote: > Conference needs a dial up DID to remind us of the AOL days!!! > > Sent from my iPhone > > On Jan 18, 2013, at 4:17 PM, Ken Rice wrote: > >> Friday Free For All is Go Get in here... >> >> SIP:888 at conference.freeswitch.org >> >> Or the normal dialup methods >> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/9c8bd51b/attachment.html From bdfoster at endigotech.com Sat Jan 19 03:51:20 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 18 Jan 2013 19:51:20 -0500 Subject: [Freeswitch-users] Friday Free For All is Go In-Reply-To: References: Message-ID: <795B4931-BD8C-4E6E-8254-73C4BC9257E3@endigotech.com> Dial up as in modem lol Sent from my iPhone On Jan 18, 2013, at 7:15 PM, Ken Rice wrote: > The conference has several DIDs on it... > > I think +1.919.386.9900 is one of them > > On 1/18/13 5:42 PM, "Brian Foster" wrote: > > Conference needs a dial up DID to remind us of the AOL days!!! > > Sent from my iPhone > > On Jan 18, 2013, at 4:17 PM, Ken Rice wrote: > > Friday Free For All is Go Get in here... > > SIP:888 at conference.freeswitch.org > > Or the normal dialup methods > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/f7138606/attachment.html From schoch+freeswitch.org at xwin32.com Sat Jan 19 04:40:00 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 18 Jan 2013 17:40:00 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: <20130116025242.83fde4e4@mail.tritonwest.net> References: <20130116025242.83fde4e4@mail.tritonwest.net> Message-ID: On Tue, Jan 15, 2013 at 6:52 PM, Dave R. Kompel wrote: > ** > Another option would be on the Linux box, enable SPANDSP modems, and use > com0com on windows to backup com ports to raw TCP connections. It supports > both. That way it would talk over a TCP pipe to an emulated modem on your > other FreeSWITCH box. > I just tried that with limited success. I turned on 2 of the SPANDSP modems (and figured I had to give the freeswitch account (the user that runs freeswitch) write permission to the /dev directory so it could create the symlinks). I then installed com0com on my Windows 7 box, installed sercd on the FS server (Linux), and ran hub4com on the Windows box to connect to sercd. As a result of all that, I can use Putty to connect do the CNCA0 serial device, and issue AT commands, and it responds like a modem! I issued an ATDT command and the call came through to the fax machine. (This is the FAX machine connected to the POTS line.) However, when I installed a "Standard 56000 bps Modem" connected to CNCA0, and attempted to use it to send a FAX from Windows, it said "An unexpected problem occurred." To make sure Freeswitch can actually do FAX, I ran originate sofia/gateway// &txfax(/path_to_fax_file) where the fax file specified was a TIFF file that I had lying around. This was successful. I'm now considering installing the Windows version of FreeSwitch on the machine of everybody who needs to send FAXes. That may take less time and trouble. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/d13a0a19/attachment-0001.html From brian at freeswitch.org Sat Jan 19 06:56:17 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Jan 2013 21:56:17 -0600 Subject: [Freeswitch-users] Friday Free For All is Go In-Reply-To: <795B4931-BD8C-4E6E-8254-73C4BC9257E3@endigotech.com> References: <795B4931-BD8C-4E6E-8254-73C4BC9257E3@endigotech.com> Message-ID: <7B264347-DB13-42CA-9BEB-3C3B8344A483@freeswitch.org> Like OMG, thats so retro! -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 ISN: 410*543 On Jan 18, 2013, at 6:51 PM, Brian Foster wrote: > Dial up as in modem lol > > Sent from my iPhone -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/546ed2d8/attachment.html From brian at freeswitch.org Sat Jan 19 06:57:43 2013 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Jan 2013 21:57:43 -0600 Subject: [Freeswitch-users] Unloop extension In-Reply-To: References: <20130117100537.56256009@mail.btcom.kz> Message-ID: <70A571F7-CB81-47A6-96E9-6B3FAB69E24E@freeswitch.org> OK lets put this to rest. 1. the reason is this. You can global_setvar to change the value without a reloadxml 2. Lets not beat a dead horse... -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 ISN: 410*543 On Jan 18, 2013, at 4:56 PM, Michael Collins wrote: > Yes, that needs to be rephrased. It really should be ${var} not $${var} since $${var} is meant to address a different scenario. > -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130118/64b84dae/attachment-0001.html From vbvbrj at gmail.com Sat Jan 19 10:38:53 2013 From: vbvbrj at gmail.com (Mimiko) Date: Sat, 19 Jan 2013 09:38:53 +0200 Subject: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 In-Reply-To: <1C7BD7EC-8865-40EE-A94D-5D2D785B99E4@gmail.com> References: <50F909AD.2010308@gmail.com> <50F91490.7080006@gmail.com> <50F9A0C0.8050401@gmail.com> <1C7BD7EC-8865-40EE-A94D-5D2D785B99E4@gmail.com> Message-ID: <50FA4D8D.4000405@gmail.com> On 19.01.2013 00:10, Steven Ayre wrote: > They should go to the cwd, get the FS process and find it by 'ls -ld /proc/PID/cwd' > > Also check the ulimit settings, -c limits the size of coredumps. > > Steve on iPhone Yep, I know the directory where FS is running. But FS user does not have write permission to that directory. "ulimit" is also set as per wiki. -- Mimiko desu. From a.afzali2003 at gmail.com Sat Jan 19 14:57:35 2013 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sat, 19 Jan 2013 15:27:35 +0330 Subject: [Freeswitch-users] Accessing SIP Replaces Header Message-ID: Hi Guys, I'm receiving INVITE message which contains Replaces header inserted by Cisco gateway. I need to be able to read it from some channel variable but I could not find in any of the my channel variables! BEST, Afshin Afzali -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130119/75df7c5e/attachment.html From gabe at gundy.org Sat Jan 19 22:12:53 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 19 Jan 2013 12:12:53 -0700 Subject: [Freeswitch-users] Accessing SIP Replaces Header In-Reply-To: References: Message-ID: On Sat, Jan 19, 2013 at 4:57 AM, afshin afzali wrote: > I'm receiving INVITE message which contains Replaces header inserted by > Cisco gateway. I need to be able to read it from some channel variable but I > could not find in any of the my channel variables! When you say you couldn't find it, what method were you using? Gabe From drk at drkngs.net Sat Jan 19 22:19:43 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Sat, 19 Jan 2013 11:19:43 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: Message-ID: <20130119191943.ca09ab6d@mail.tritonwest.net> I use the windows version of FreeSWITCH with fax modems on a Windows Server box, running the normal Microsoft Fax Service. It's been working using the SPANDSP fax modems perfict for over a year. The thing you have to remember is that the modem in mod_spandsp is T.31, not T.38. If you wan't T.38 you need to run the call through T.38 Gateway, just as if the modem device as a TDM line. Look at the SPANDSP wiki page for reference. I know this works on windows, and should also work on linux. I'm not sure about the built in client fax in windows 7, but I know it works with the fax service role on windows server. --Dave _____ From: Steven Schoch [mailto:schoch+freeswitch.org at xwin32.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Fri, 18 Jan 2013 17:40:00 -0800 Subject: Re: [Freeswitch-users] FAX problems On Tue, Jan 15, 2013 at 6:52 PM, Dave R. Kompel wrote: Another option would be on the Linux box, enable SPANDSP modems, and use com0com on windows to backup com ports to raw TCP connections. It supports both. That way it would talk over a TCP pipe to an emulated modem on your other FreeSWITCH box. I just tried that with limited success. I turned on 2 of the SPANDSP modems (and figured I had to give the freeswitch account (the user that runs freeswitch) write permission to the /dev directory so it could create the symlinks). I then installed com0com on my Windows 7 box, installed sercd on the FS server (Linux), and ran hub4com on the Windows box to connect to sercd. As a result of all that, I can use Putty to connect do the CNCA0 serial device, and issue AT commands, and it responds like a modem! I issued an ATDT command and the call came through to the fax machine. (This is the FAX machine connected to the POTS line.) However, when I installed a "Standard 56000 bps Modem" connected to CNCA0, and attempted to use it to send a FAX from Windows, it said "An unexpected problem occurred." To make sure Freeswitch can actually do FAX, I ran originate sofia/gateway// &txfax(/path_to_fax_file) where the fax file specified was a TIFF file that I had lying around. This was successful. I'm now considering installing the Windows version of FreeSwitch on the machine of everybody who needs to send FAXes. That may take less time and trouble. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130119/09f34847/attachment.html From craig.robert at hotmail.com Sat Jan 19 22:54:55 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Sat, 19 Jan 2013 12:54:55 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com> References: , , , , , , , , , , , , , <50F975E5.2040206@coppice.org>, , , , <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com> Message-ID: OK, I copied everything from the fax extension dial plan into the below pastebin. http://pastebin.freeswitch.org/20466 Robert From: bdfoster at endigotech.com Date: Fri, 18 Jan 2013 12:59:38 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Submit the dialplan you are using to receive the fax. You can use the FreeSWITCH pastebin, http://pastebin.freeswitch.org Sent from my iPhone On Jan 18, 2013, at 12:43 PM, Robert Craig wrote: I believe I'm just using the built in SpanDSP for faxing. I've looked around for 'rxfax' and can't find anything. I did notice that the directory where the faxes were supposed to be stored for the extension I create didn't even exist. I created them, applied the permissions, but it seems like I am missing something because the same error is still there. Robert Date: Fri, 18 Jan 2013 09:13:36 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration If you are using the rxfax app then you specify the file name and location: Make sure that the target directory has read/write permissions. You can use something like "chmod 666 /my/directory" to make it read/write. -MC On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: Where does the file get downloaded to? What should the permissions be? Robert > Date: Sat, 19 Jan 2013 00:18:45 +0800 > From: steveu at coppice.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If the TIFF file cannot be opened on receive you probably have a file > permissions issue (e.g. with the directory in which the file is to be > created). If the TIFF file cannot be opened on transmit there are > various possibilities about the suitability of the file's contents. > > Steve > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > This was an incoming fax. I don't have much control over the other > > end. Why is that error appearing when trying to receive something? > > > > Robert > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > wrote: > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > >> Please try pdf, you should be in business now. > >> Please let us know, > >> > >> On Jan 18, 2013 12:35 AM, "Robert Craig" >> > wrote: > >> > >> OK, so I've made a little bit of progress. The distant end is a > >> Windows Fax server hooked up to an analog line. I created a fax > >> extension in FS, assigned an inbound route to the extension, and > >> made sure the extension had an email address. I believe the fax > >> receive wants to complete, but there is an interesting error in > >> the log about TIF files. Anyone ever seen this before? > >> > >> v=0 > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> c=IN IP4 72.201.112.131 > >> t=0 0 > >> m=image 27714 udptl t38 > >> a=T38FaxVersion:0 > >> a=T38MaxBitRate:9600 > >> a=T38FaxFillBitRemoval > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxUdpEC:t38UDPRedundancy > >> > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > >> ip/port confirmed. > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > >> Negotiation Result > >> ======================================================= > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > >> status off > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > >> country: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > >> vendor: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > >> model: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > >> processing not successful - result (41) TIFF/F file cannot be > >> opened.* > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > >> transferred: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > >> fax pages: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > >> resolution: 0x0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > >> status off > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > >> country: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > >> vendor: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > >> model: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > >> sofia/external/+14792460547 @flowroute.com > >> Restore previous codec PCMU:0. > >> EXECUTE sofia/external/+14792460547 > >> @flowroute.com hangup() > >> > >> Robert > >> > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> OK, I am not extremely concerned with getting an ATA working > >> right now. If I could just get an inbound extension-to-email > >> working it would be great. I'll start from scratch and then let > >> you know what happens. Just to make sure, I am going to create an > >> extension, assign an email address to it, make sure there is an > >> inbound route to that extension, and then test it. > >> > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > >> From: gassaad at emassembly.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> The reason for outbound extension is: > >> When you send a fax from ATA to the outside world, it will dial > >> first to the outbound extension and then after successful receive > >> it will dial out to the intended number. > >> > >> As for the error you have it may be related to your ATA. I am not > >> sure please see why do you get the following message. > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > >> > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > >> > wrote: > >> > >> OK, I've already tried creating one extension for inbound and > >> set an email address. Of course, it failed and the errors are > >> at the bottom of this email. What is the purpose of the > >> outbound extension? > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > >> From: gassaad at emassembly.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Sorry for the delay to get back to you. > >> 1- App --> fax server > >> 2- Create two extensions (one for inbound) another for > >> (outbound) From within either extension you can send faxes > >> "virtual fax" > >> 3- point your create your outbound and inbound dialplans as > >> normal but make it to point to those extensions. > >> > >> Good luck, > >> > >> George > >> > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > >> > > >> wrote: > >> > >> What did you do on the config side to get it to work? > >> I've tried both an Ata and a fax to email extension and > >> they both fail in both directions. > >> > >> Robert > >> > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > >> > > >> wrote: > >> > >> I use Fusion successfully to send / receive fax. > >> The way it works: > >> ATA --->FaxServer. > >> After successful receipt, Fax Server---> DID where > >> you have dialled originally. > >> > >> Cheers, > >> > >> George > >> > >> > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > >> >> > wrote: > >> > >> Did anyone see the issue or what I may have > >> missed in the config? As mentioned earlier, I > >> thought it would after a fresh install. Any help > >> is appreciated. > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Thanks for the quick response. Below is the > >> capture from the log viewer. In this test, I am > >> using a remote fax machine (line from Cox with > >> physical machine) faxing into the "email server" > >> where I created an extension for fax-to-email. > >> Calling number is 1479-246-0547. > >> > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:99 > >> sofia/external/+14792460547 at flowroute.com > >> Standard DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> mod_sofia.c:396 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:559 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State Change > >> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:556 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate Change HANGUP > >> -> DOWN > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1510 Close Channel > >> sofia/external/+14792460547 at flowroute.com > >> [CS_DESTROY] > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1506 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Ended > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1488 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Locked, Waiting on > >> external entities > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:440 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change CS_REPORTING > >> -> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State > >> REPORTING going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:92 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> REPORTING, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:446 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_HANGUP -> CS_REPORTING > >> > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:599 Hangup Command > >> with no Session system(/usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 > >> name=+14792460547-2013-01-10-13-39-26 > >> messages='result: Disconnected after permitted > >> retries sender:ATW-4792460547 pages:0' > >> domain=192.168.4.15 caller_id_name='' > >> caller_id_number=): > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:48 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> HANGUP, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:552 Sending BYE to > >> sofia/external/+14792460547 at flowroute.com > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:503 Channel > >> sofia/external/+14792460547 at flowroute.com > >> hanging up, > >> cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:2687 > >> sofia/external/+14792460547 at flowroute.com > >> skip receive > >> message [APPLICATION_EXEC_COMPLETE] (channel is > >> hungup already) > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:3017 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [KILL] > >> 2013-01-10 13:39:52.216151 [NOTICE] > >> mod_dptools.c:1150 Hangup > >> sofia/external/+14792460547 at flowroute.com > >> [CS_EXECUTE] > >> [NORMAL_CLEARING] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:2994 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change ACTIVE -> HANGUP > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> hangup() > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_codec.c:244 > >> sofia/external/+14792460547 at flowroute.com > >> Restore > >> previous codec PCMU:0. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:516 remote model: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:515 remote vendor: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:514 remote country: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:513 ECM status off > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:509 Total fax pages: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:507 Pages transferred: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > >> Fax Ident > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:505 Remote station id: > >> ATW-4792460547 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:500 Fax processing not > >> successful - result (48) Disconnected after > >> permitted retries. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> 2013-01-10 13:39:38.766190 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:29.576127 [ERR] > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > >> must be between 20 and 10000 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxRateManagement:transferredTCF > >> m=image 31632 udptl t38 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completing][200] > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > >> sofia/external/+14792460547 at flowroute.com > >> Update Callee > >> ID to "+14792460547" <+14792460547> > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [calling][0] > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:2647 Local SDP: > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1379 Raw write codec activation > >> Success L16 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_core_codec.c:219 > >> sofia/external/+14792460547 at flowroute.com > >> Push codec L16:70 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1363 Raw read codec activation > >> Success L16 20000 > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > >> > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_ivr_play_say.c:1682 done playing file > >> silence_stream://2000 > >> 2013-01-10 13:39:26.256197 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> switch_ivr_play_say.c:1309 Codec Activated > >> L16 at 8000hz 1 channels 20ms > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> playback(silence_stream://2000) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38_request]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38_request=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38]=[true] > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completed][200] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.006127 [NOTICE] > >> mod_dptools.c:1176 Channel > >> [sofia/external/+14792460547 at flowroute.com > >> ] has been > >> answered > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_channel.c:3395 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change RINGING -> ACTIVE > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=sendrecv > >> a=ptime:20 > >> a=silenceSupp:off - - - - > >> a=fmtp:101 0-16 > >> a=rtpmap:101 telephone-event/8000 > >> a=rtpmap:0 PCMU/8000 > >> m=audio 27588 RTP/AVP 0 101 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_sofia.c:856 Local SDP > >> sofia/external/+14792460547 at flowroute.com > >> : > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:1046 STUN Success > >> [72.201.112.131]:[27588] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> answer() > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [api_hangup_hook]=[system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=${last_fax} messages='result: > >> ${fax_result_text} > >> sender:${fax_remote_station_id} > >> pages:${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='${caller_id_name}' > >> caller_id_number=${caller_id_number}] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\${last_fax} > >> messages='result: \${fax_result_text} > >> sender:\${fax_remote_station_id} > >> pages:\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\${caller_id_name}' > >> caller_id_number=\${caller_id_number}) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [call_direction]=[local] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(call_direction=local) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action hangup() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> playback(silence_stream://2000) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38_request=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action answer() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\\\${last_fax} > >> messages='result: \\\${fax_result_text} > >> sender:\\\${fax_remote_station_id} > >> pages:\\\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\\\${caller_id_name}' > >> caller_id_number=\\\${caller_id_number}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test Fax] destination_number(3150) =~ > >> /^3150$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Test Fax] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain_user] destination_number(3150) =~ > >> /^\*97$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain_user] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain] destination_number(3150) =~ > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [send_to_voicemail] > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->send_to_voicemail] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [extension-intercom] > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->extension-intercom] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [intercept-ext] destination_number(3150) > >> =~ /^\*\*(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->intercept-ext] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_return] destination_number(3150) =~ > >> /^\*69$|^lcr$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_return] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_privacy] destination_number(3150) =~ > >> /^\*67(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_privacy] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [global] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > >> break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->global] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [redial] destination_number(3150) =~ > >> /^(redial|\*870)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->redial] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [group-intercept] destination_number(3150) > >> =~ /^\*8$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->group-intercept] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> ANTI-Action > >> set(call_direction=local) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call-direction] ${call_direction}() =~ > >> /^(inbound|outbound|local)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call-direction] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [Flowroute.11d] destination_number(3150) > >> =~ /^\+?(\d{11})$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Flowroute.11d] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->3150 in context default > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:25.816280 [NOTICE] > >> switch_ivr.c:1781 Transfer > >> sofia/external/+14792460547 at flowroute.com > >> to > >> XML[3150 at default] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_ivr.c:1774 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_EXECUTE -> CS_ROUTING > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> transfer(3150 > >> XML default) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_channel.c:1135 EXPORT (export_vars) > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [outside_call]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(outside_call=true) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> transfer(3150 XML default) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] > >> destination_number(15203039810) =~ > >> /^15203039810$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] context(public) =~ /public/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->Test_Number] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_debug] ${call_debug}(false) =~ > >> /^true$/ break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->call_debug] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(outside_call=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [outside_call] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->outside_call] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->unloop] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->15203039810 in context public > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_channel.c:2003 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change DOWN -> RINGING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:454 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State INIT > >> going to sleep > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> mod_sofia.c:126 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_INIT -> CS_ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:454 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_NEW -> CS_INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5263 Set 2833 dtmf send/recv payload > >> to 101 > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_codec.c:111 > >> sofia/external/+14792460547 at flowroute.com > >> Original read > >> codec set to PCMU:0 > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:3093 Set Codec > >> sofia/external/+14792460547 at flowroute.com > >> PCMU/8000 20 > >> ms 160 samples 64000 bits > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5134 Audio Codec Compare > >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5134 Audio Codec Compare > >> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > >> > >> a=ptime:20 > >> a=fmtp:101 0-15 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:18 annexb=no > >> a=rtpmap:18 G729/8000 > >> m=audio 31632 RTP/AVP 0 18 101 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849875 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [received][100] > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_state_machine.c:433 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State NEW > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_NEW > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.786370 [NOTICE] > >> switch_channel.c:968 New Channel > >> sofia/external/+14792460547 at flowroute.com > >> > >> [d276f012-5b65-11e2-beae-8564ff827a25] > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Thu, 10 Jan 2013 09:47:18 -0800 > >> From: msc at freeswitch.org > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Can you pastebin a console debug log w/ sip trace > >> of a call failing to negotiate t.38 correctly? > >> That might help us figure out where to look. > >> -MC > >> > >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig > >> >> > wrote: > >> > >> Just installed FusionPBX using the script > >> built for a Centos 6 install. Everything > >> works great in regards to calls, but my main > >> goal of this project was to utilize the fax > >> capabilities of Freeswitch/FusionPBX. I have > >> tried many things in media bypass, routing to > >> a fax extension, using an ATA, etc., but I > >> can't get T38 to negotiate correctly. Can > >> someone point me in the right direction as to > >> where I should begin? I am guessing it isn't > >> supposed to work right out of the box. I am > >> using Flowroute as my provider right now and > >> the have confirmed that they utilize T38 in > >> both directions. Thanks! > >> > >> > >> Robert > >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130119/0ad1398a/attachment-0001.html From rnbrady at gmail.com Sun Jan 20 02:08:58 2013 From: rnbrady at gmail.com (Richard Brady) Date: Sat, 19 Jan 2013 23:08:58 +0000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> <20130118102614.GA27116@eagle.cupis.co.uk> <8A80C99638A5E24E83696EAE33A9B6B968B5302824@twbexmbx001.ecn.hosting> Message-ID: Hey Mark A few thoughts: 1. If you are using start_dtmf on the FS box then you are making it do media processing which is fine but this is a change from what the OpenSER / RTPproxy were doing for you. If you want it to be a drop-in replacement you might want to look at some of the following: proxy_media inbound-proxy-media http://wiki.freeswitch.org/wiki/Proxy_Media disable-transcoding codec_string / ep_codec_string inherit_codec http://wiki.freeswitch.org/wiki/Codec_negotiation 2. It's dtmf-type in the profile or dtmf_type in the dialplan. Not dtmf_mode as you mention in an earlier message. 3. Your post mentioning PT 18 and the annexb parameter suggests you are using G729, which does not support inband DTMF reliably, if at all. 4. You will need to put the sip_append_audio_sdp before the pre_answer (which you should get rid off as suggested already). 5. if you do want FS to convert from inband to rfc2833, there was a previous post suggesting spandsp_start_dtmf may perform better than start_dtmf (although I can't verify this). Regards, Richard On 18 January 2013 16:12, Steven Ayre wrote: > Did you try removing the rfc2833-pt param and setting name="dtmf-type" value="none"/> on the outgoing sofia profile to the PBX? > > Inband tones will always be passed over, FS would actively need to detect > the tone and remove it from the media to avoid doing so. So the PBX is not > looking inband for it. Since there's no way in SIP to notify that inband > tones are in use the PBX is probably checking for out-of-band DTMF and not > checking inband at all if out-of-band has been negotiated - so what you > probably need to is disable out of band media entirely. > > As far as comparing FS behaviour to OpenSER goes I would avoid it. They're > very different products. Understand the difference between a SIP proxy > (OpenSER) and a B2BUA (FreeSWITCH). OpenSER only forwards SIP messages with > media terminating on the endpoints. FS negotiates 2 separate calls > including media and terminates both calls on FS and passes media between > them. As such OpenSER doesn't do any media negotiation and FS does which > will make their behaviour rather different. > > -Steve > > > > > > On 18 January 2013 15:20, support at ecn.net.au wrote: > >> Yes, we read that and tried to set that when attempting multiple >> configurations (attempting to pass inband dtmf). >> >> Currently we can't get Freeswitch in an SBC role to pass the inband in >> the media stream at all (see earlier email) - also having odd issues with >> start_dtmf when accessing Asterisk IVR's (again very odd). the only >> difference we can see (between FS and our legacy openser) is the Invite >> pushed on the bleg on Openser included the fmtp:18 annexb=no directive (as >> per the telco's SDP to us); however on FS it doesn't forward these (and we >> can't seem to force FS to add these attributes to the SDP on the bleg). >> >> Again the start_dtmf issues exists only in 1.2 and 1.3 FS , in the 1.0.6 >> the issue is not present (is this a bug possibly?) >> >> Any ideas? >> >> Kind Regards, >> >> Mark >> >> ________________________________________ >> From: Paul Cupis [paul at cupis.co.uk] >> Sent: Friday, 18 January 2013 8:26 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] SBC In-band DTMF >> >> On Fri, Jan 18, 2013 at 01:57:36AM +0000, Steven Ayre wrote: >> > >> > >> > What're you hoping to do here? This sets the payload type value. 0 >> is a >> > valid pt. This won't disable it, it'll instead use the value 0. That >> > happens to be the one for G711, so if that codec is enabled you're >> just >> > going to get a conflict. >> >> If the rfc2833-pt is set to a value <96 it is outside the allowed range >> for dynamic payload negotiation and FS will not offer RFC2833 support >> in the SDP. >> >> ref: sofia_glue.c >> >> Regards, >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130119/280da77a/attachment.html From mario_fs at mgtech.com Sun Jan 20 03:06:57 2013 From: mario_fs at mgtech.com (Mario G) Date: Sat, 19 Jan 2013 16:06:57 -0800 Subject: [Freeswitch-users] adding In-Reply-To in sip header not working Message-ID: I never did this so I must be missing something, I tried both below but the bridge then fails. Can anyone shed some light on what I am doing wrong. My ITSP now supports in-reply-to so I can pass the caller ID to a forwarded call from FS. Mario G ]]> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130119/a87e1af4/attachment.html From rnbrady at gmail.com Sun Jan 20 03:11:50 2013 From: rnbrady at gmail.com (Richard Brady) Date: Sun, 20 Jan 2013 00:11:50 +0000 Subject: [Freeswitch-users] FS hangs up after 30 seconds by timeout on ACK (after sending 200 OK) In-Reply-To: References: Message-ID: Hi there > When someone calls to num_msk, it shoot to port 5070. Call is received, > but is rejected after 30 seconds by timeout (not received ACK on 200 OK). I > think it's because FS sends 200 OK with port number = 0 in field Contact. > All another headers are good (it seems). > In the profile check the value of ext-sip-ip is not stun or auto-nat as this could result in the 0 port number if the discovery fails (just a guess on my part). Better to set it to auto or the actual IP address. > Does someone know how to say FS to use correct port number in field > Contact in 200 OK? > If the suggestion above does not work there is an undocumented profile parameter of ext-sip-port so you could try that. Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/241424ec/attachment.html From rnbrady at gmail.com Sun Jan 20 03:27:40 2013 From: rnbrady at gmail.com (Richard Brady) Date: Sun, 20 Jan 2013 00:27:40 +0000 Subject: [Freeswitch-users] FS hangs up after 30 seconds by timeout on ACK (after sending 200 OK) In-Reply-To: References: Message-ID: PS: Make sure you're using an up to date version and and see http://jira.freeswitch.org/browse/FS-4859 On 20 January 2013 00:11, Richard Brady wrote: > Hi there > > >> When someone calls to num_msk, it shoot to port 5070. Call is received, >> but is rejected after 30 seconds by timeout (not received ACK on 200 OK). I >> think it's because FS sends 200 OK with port number = 0 in field Contact. >> All another headers are good (it seems). >> > > In the profile check the value of ext-sip-ip is not stun or auto-nat as > this could result in the 0 port number if the discovery fails (just a guess > on my part). Better to set it to auto or the actual IP address. > > >> Does someone know how to say FS to use correct port number in field >> Contact in 200 OK? >> > > If the suggestion above does not work there is an undocumented profile > parameter of ext-sip-port so you could try that. > > Richard > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/fe872aea/attachment.html From lists at kavun.ch Sun Jan 20 03:31:57 2013 From: lists at kavun.ch (Emrah) Date: Sat, 19 Jan 2013 19:31:57 -0500 Subject: [Freeswitch-users] Cannot save Voicemail message In-Reply-To: References: <3263505F-2AC8-4BB5-ABEB-44B706AD80C5@kavun.ch> Message-ID: <6770CA93-7D95-404A-809A-732A68C60771@kavun.ch> The console logs are not very helpful? http://pastebin.freeswitch.org/20467 Thanks for any help. E On Jan 18, 2013, at 6:24 PM, Michael Collins wrote: > Can you supply a bit more context? Show us more log lines. > -MC > > On Fri, Jan 18, 2013 at 2:18 PM, Emrah wrote: > Hi all, > > I've noticed a strange problem lately, I can't press 3 to save a VM. > > I use mod_shout, my VMs are in MP3 format. > > Log abstract: > > 2013-01-18 17:11:09.191700 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 16000hz 1 channels 20ms > 2013-01-18 17:11:10.191704 [DEBUG] switch_rtp.c:3827 RTP RECV DTMF 3:1120 > 2013-01-18 17:11:10.191704 [DEBUG] switch_ivr_play_say.c:1682 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-save_recording.wav > 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] > 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-you_have.wav] (en:en) > > freeswitch at internal> version > FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z) > > Deleting works fine. > Any idea? > > Thanks and all the best, > Emrah > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rnbrady at gmail.com Sun Jan 20 04:13:42 2013 From: rnbrady at gmail.com (Richard Brady) Date: Sun, 20 Jan 2013 01:13:42 +0000 Subject: [Freeswitch-users] adding In-Reply-To in sip header not working In-Reply-To: References: Message-ID: On 20 January 2013 00:06, Mario G wrote: > I never did this so I must be missing something, I tried both below but > the bridge then fails. Can anyone shed some light on what I am doing wrong. > My ITSP now supports in-reply-to so I can pass the caller ID to a forwarded > call from FS. > In-Reply-To should contain a Call-ID not a caller ID. They are very different. The following would make a bit more sense, but still not a lot: Using In-Reply-To in a response doesn't seem right to me. I would expect it to appear in an INVITE. For for example, you get a missed call and you call the person back, then the INVITE for the callback would have a new Call-ID but the original Call-ID in the In-Reply-To header. That said, I have no idea what your ITSPs intended use for the header is. Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/9fa7e6c2/attachment.html From a.afzali2003 at gmail.com Sun Jan 20 08:44:52 2013 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sun, 20 Jan 2013 09:14:52 +0330 Subject: [Freeswitch-users] Accessing SIP Replaces Header In-Reply-To: References: Message-ID: Hi Gabe, Actually I expect to have a variable such as sip_replaces / variable_sip_replaces ! but I don't have any :( On Sat, Jan 19, 2013 at 10:42 PM, Gabriel Gunderson wrote: > On Sat, Jan 19, 2013 at 4:57 AM, afshin afzali > wrote: > > I'm receiving INVITE message which contains Replaces header inserted by > > Cisco gateway. I need to be able to read it from some channel variable > but I > > could not find in any of the my channel variables! > > When you say you couldn't find it, what method were you using? > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/7e74efce/attachment.html From a.afzali2003 at gmail.com Sun Jan 20 08:50:02 2013 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sun, 20 Jan 2013 09:20:02 +0330 Subject: [Freeswitch-users] Accessing SIP Replaces Header In-Reply-To: References: Message-ID: I use Lua's session:getVariable to get it, also using uuid_dump to list all available variables. Afshin On Sun, Jan 20, 2013 at 9:14 AM, afshin afzali wrote: > Hi Gabe, > > Actually I expect to have a variable such as sip_replaces / > variable_sip_replaces ! but I don't have any :( > > > > > On Sat, Jan 19, 2013 at 10:42 PM, Gabriel Gunderson wrote: > >> On Sat, Jan 19, 2013 at 4:57 AM, afshin afzali >> wrote: >> > I'm receiving INVITE message which contains Replaces header inserted by >> > Cisco gateway. I need to be able to read it from some channel variable >> but I >> > could not find in any of the my channel variables! >> >> When you say you couldn't find it, what method were you using? >> >> >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/c853e44a/attachment.html From jtbock at synacktics.com Sun Jan 20 09:14:05 2013 From: jtbock at synacktics.com (Tim Bock) Date: Sat, 19 Jan 2013 23:14:05 -0700 Subject: [Freeswitch-users] Digium TDM410 + DTMF + echo cancel Message-ID: <50FB8B2D.6060502@synacktics.com> Hi, I am having a continuing problem with freeswitch and the Digium TDM card. For some period of time after a reboot, DTMF is detected correctly for incoming calls, and there is no problem with echo. All is good. Then at some point, dtmf is no longer detected on incoming calls handled by the TDM card. Presumably at the same time that dtmf quits being detected, the echo cancellation quits working. Again, just on incoming calls handled by the Digium card. The only way I've been able to get things working again is to reboot the server, but it doesn't seem to last for long. Can anyone suggest how I might track down what is wrong? Thanks, Tim From charlie at charlie.is Sat Jan 19 22:56:38 2013 From: charlie at charlie.is (Charlie Orford) Date: Sat, 19 Jan 2013 20:56:38 +0100 Subject: [Freeswitch-users] mod_voicemail always sends empty emails Message-ID: <50FAFA76.3010708@charlie.is> Hi list, Voice mails get correctly saved in /usr/local/freeswitch/storage/voicemail/default/domain.com/extnum/ but the email sent by mod_voicemail always arrives blank with the wav file attachment missing. To check the problem wasn't mail server related, I setup a simple postfix "Local only" installation on the same host as FS. Here's the entire contents of a voicemail email sent to postfix by FS (note: domain.com is substituted for real domain): # cat /var/mail/root From voicemail at domain.com Sat Jan 19 17:38:03 2013 Return-Path: X-Original-To: postmaster at fs.domain.com Delivered-To: postmaster at fs.domain.com Received: by fs (Postfix, from userid 1001) id 9E6AC23E57; Sat, 19 Jan 2013 17:38:03 +0000 (UTC) Message-Id: <20130119183349.9E6AC23E57 at fs> Date: Sat, 19 Jan 2013 18:33:49 +0000 (UTC) From: voicemail at angloeuropean.com To: undisclosed-recipients:; And here are the voicemail specific config params for the extension: /usr/local/freeswitch/conf/voicemail.tpl exists and looks correct (contents here: http://pastebin.com/Tr4fvyht ) I took a log (via fsi_cli and /log 7) of a call leaving a voicemail to that extension (here: http://pastebin.com/B1zteGV7 ) but nothing jumps out. Is there anything else I can do to track down what's causing this? I'm running the latest stable branch on Debian squeeze: FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z) Thanks, Charlie From slava at tangramltd.com Sun Jan 20 14:24:58 2013 From: slava at tangramltd.com (Dubrovskiy Viacheslav) Date: Sun, 20 Jan 2013 13:24:58 +0200 Subject: [Freeswitch-users] Digium TDM410 + DTMF + echo cancel In-Reply-To: <50FB8B2D.6060502@synacktics.com> References: <50FB8B2D.6060502@synacktics.com> Message-ID: <50FBD40A.8080109@tangramltd.com> 20.01.2013 08:14, Tim Bock ?????: > Hi, > > I am having a continuing problem with freeswitch and the Digium TDM > card. For some period of time after a reboot, DTMF is detected > correctly for incoming calls, and there is no problem with echo. All is > good. > > Then at some point, dtmf is no longer detected on incoming calls handled > by the TDM card. Presumably at the same time that dtmf quits being > detected, the echo cancellation quits working. Again, just on incoming > calls handled by the Digium card. The only way I've been able to get > things working again is to reboot the server, but it doesn't seem to > last for long. > > Can anyone suggest how I might track down what is wrong? Try enable DTMF debug in freetdm and compare results. Example for freetdm.conf [general] debugdtmf_directory=/var/spool/freeswitch/dtmf [span zt mySpan] debugdtmf => yes -- WBR, Dubrovskiy Viacheslav -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3746 bytes Desc: ?????????????????????????????????? ?????????????? S/MIME Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/d50c3270/attachment-0001.bin From royj at yandex.ru Sun Jan 20 17:16:03 2013 From: royj at yandex.ru (royj) Date: Sun, 20 Jan 2013 18:16:03 +0400 Subject: [Freeswitch-users] video calls, full processing media mode Message-ID: <20130120181603.d6c2880b8bf3f7c42f03fded@yandex.ru> Is there ability to establish video call through FreeSWITCH with full processing media mode on profile. I see FreeSWITCH answers 183 Session Progress with 'm=video 0 RTP/AVP 19.', when video-phone offers for example 'm=video 4002 RTP/AVP 97.' -- Regards, royj From gassaad at emassembly.com Sun Jan 20 17:52:11 2013 From: gassaad at emassembly.com (George Assaad) Date: Sun, 20 Jan 2013 09:52:11 -0500 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: <50F975E5.2040206@coppice.org> <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com> Message-ID: Here is mine: Inbound: http://pastebin.freeswitch.org/20471 Outbound: http://pastebin.freeswitch.org/20470 Thanks, George On Sat, Jan 19, 2013 at 2:54 PM, Robert Craig wrote: > OK, I copied everything from the fax extension dial plan into the below > pastebin. > > http://pastebin.freeswitch.org/20466 > > > Robert > > > ------------------------------ > From: bdfoster at endigotech.com > Date: Fri, 18 Jan 2013 12:59:38 -0500 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Submit the dialplan you are using to receive the fax. You can use the > FreeSWITCH pastebin, http://pastebin.freeswitch.org > > Sent from my iPhone > > On Jan 18, 2013, at 12:43 PM, Robert Craig > wrote: > > I believe I'm just using the built in SpanDSP for faxing. I've looked > around for 'rxfax' and can't find anything. I did notice that the directory > where the faxes were supposed to be stored for the extension I create > didn't even exist. I created them, applied the permissions, but it seems > like I am missing something because the same error is still there. > > > Robert > > > ------------------------------ > Date: Fri, 18 Jan 2013 09:13:36 -0800 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If you are using the rxfax app then you specify the file name and location: > > > > Make sure that the target directory has read/write permissions. You can > use something like "chmod 666 /my/directory" to make it read/write. > > -MC > > On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: > > Where does the file get downloaded to? What should the permissions be? > > Robert > > > > Date: Sat, 19 Jan 2013 00:18:45 +0800 > > From: steveu at coppice.org > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] T38 Configuration > > > > If the TIFF file cannot be opened on receive you probably have a file > > permissions issue (e.g. with the directory in which the file is to be > > created). If the TIFF file cannot be opened on transmit there are > > various possibilities about the suitability of the file's contents. > > > > Steve > > > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > > This was an incoming fax. I don't have much control over the other > > > end. Why is that error appearing when trying to receive something? > > > > > > Robert > > > > > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > > wrote: > > > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > > >> Please try pdf, you should be in business now. > > >> Please let us know, > > >> > > >> On Jan 18, 2013 12:35 AM, "Robert Craig" > >> > wrote: > > >> > > >> OK, so I've made a little bit of progress. The distant end is a > > >> Windows Fax server hooked up to an analog line. I created a fax > > >> extension in FS, assigned an inbound route to the extension, and > > >> made sure the extension had an email address. I believe the fax > > >> receive wants to complete, but there is an interesting error in > > >> the log about TIF files. Anyone ever seen this before? > > >> > > >> v=0 > > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> c=IN IP4 72.201.112.131 > > >> t=0 0 > > >> m=image 27714 udptl t38 > > >> a=T38FaxVersion:0 > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxFillBitRemoval > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > > >> ip/port confirmed. > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > > >> Negotiation Result > > >> ======================================================= > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > > >> status off > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > > >> country: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > > >> vendor: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > > >> model: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > > >> processing not successful - result (41) TIFF/F file cannot be > > >> opened.* > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > > >> transferred: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > > >> fax pages: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > > >> resolution: 0x0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > > >> status off > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > > >> country: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > > >> vendor: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > > >> model: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > > >> sofia/external/+14792460547 @flowroute.com > > >> Restore previous codec PCMU:0. > > >> EXECUTE sofia/external/+14792460547 > > >> @flowroute.com hangup() > > >> > > >> Robert > > >> > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> OK, I am not extremely concerned with getting an ATA working > > >> right now. If I could just get an inbound extension-to-email > > >> working it would be great. I'll start from scratch and then let > > >> you know what happens. Just to make sure, I am going to create an > > >> extension, assign an email address to it, make sure there is an > > >> inbound route to that extension, and then test it. > > >> > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > > >> From: gassaad at emassembly.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> The reason for outbound extension is: > > >> When you send a fax from ATA to the outside world, it will dial > > >> first to the outbound extension and then after successful receive > > >> it will dial out to the intended number. > > >> > > >> As for the error you have it may be related to your ATA. I am not > > >> sure please see why do you get the following message. > > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > > >> > > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > > >> > wrote: > > >> > > >> OK, I've already tried creating one extension for inbound and > > >> set an email address. Of course, it failed and the errors are > > >> at the bottom of this email. What is the purpose of the > > >> outbound extension? > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > > >> From: gassaad at emassembly.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Sorry for the delay to get back to you. > > >> 1- App --> fax server > > >> 2- Create two extensions (one for inbound) another for > > >> (outbound) From within either extension you can send faxes > > >> "virtual fax" > > >> 3- point your create your outbound and inbound dialplans as > > >> normal but make it to point to those extensions. > > >> > > >> Good luck, > > >> > > >> George > > >> > > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > > >> > > > >> wrote: > > >> > > >> What did you do on the config side to get it to work? > > >> I've tried both an Ata and a fax to email extension and > > >> they both fail in both directions. > > >> > > >> Robert > > >> > > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > > >> > > > >> wrote: > > >> > > >> I use Fusion successfully to send / receive fax. > > >> The way it works: > > >> ATA --->FaxServer. > > >> After successful receipt, Fax Server---> DID where > > >> you have dialled originally. > > >> > > >> Cheers, > > >> > > >> George > > >> > > >> > > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > > >> > >> > wrote: > > >> > > >> Did anyone see the issue or what I may have > > >> missed in the config? As mentioned earlier, I > > >> thought it would after a fresh install. Any help > > >> is appreciated. > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Thanks for the quick response. Below is the > > >> capture from the log viewer. In this test, I am > > >> using a remote fax machine (line from Cox with > > >> physical machine) faxing into the "email server" > > >> where I created an extension for fax-to-email. > > >> Calling number is 1479-246-0547. > > >> > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:99 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> mod_sofia.c:396 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:559 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State Change > > >> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:556 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate Change HANGUP > > >> -> DOWN > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1510 Close Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_DESTROY] > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1506 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Ended > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1488 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Locked, Waiting on > > >> external entities > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:440 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change CS_REPORTING > > >> -> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State > > >> REPORTING going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:92 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> REPORTING, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:446 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_HANGUP -> CS_REPORTING > > >> > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:599 Hangup Command > > >> with no Session system(/usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 > > >> name=+14792460547-2013-01-10-13-39-26 > > >> messages='result: Disconnected after permitted > > >> retries sender:ATW-4792460547 pages:0' > > >> domain=192.168.4.15 caller_id_name='' > > >> caller_id_number=): > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:48 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> HANGUP, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:552 Sending BYE to > > >> sofia/external/+14792460547 at flowroute.com > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:503 Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> hanging up, > > >> cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:2687 > > >> sofia/external/+14792460547 at flowroute.com > > >> skip receive > > >> message [APPLICATION_EXEC_COMPLETE] (channel is > > >> hungup already) > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:3017 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [KILL] > > >> 2013-01-10 13:39:52.216151 [NOTICE] > > >> mod_dptools.c:1150 Hangup > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_EXECUTE] > > >> [NORMAL_CLEARING] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:2994 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change ACTIVE -> HANGUP > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> hangup() > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_codec.c:244 > > >> sofia/external/+14792460547 at flowroute.com > > >> Restore > > >> previous codec PCMU:0. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:516 remote model: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:515 remote vendor: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:514 remote country: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:513 ECM status off > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:509 Total fax pages: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:507 Pages transferred: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > > >> Fax Ident > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:505 Remote station id: > > >> ATW-4792460547 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:500 Fax processing not > > >> successful - result (48) Disconnected after > > >> permitted retries. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:38.766190 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:29.576127 [ERR] > > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > > >> must be between 20 and 10000 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxRateManagement:transferredTCF > > >> m=image 31632 udptl t38 > > >> t=0 0 > > >> c=IN IP4 66.42.121.5 > > >> s=- > > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > > >> v=0 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > > >> Remote SDP: > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completing][200] > > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > > >> sofia/external/+14792460547 at flowroute.com > > >> Update Callee > > >> ID to "+14792460547" <+14792460547> > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [calling][0] > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:2647 Local SDP: > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1379 Raw write codec activation > > >> Success L16 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_core_codec.c:219 > > >> sofia/external/+14792460547 at flowroute.com > > >> Push codec L16:70 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1363 Raw read codec activation > > >> Success L16 20000 > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > > >> > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_ivr_play_say.c:1682 done playing file > > >> silence_stream://2000 > > >> 2013-01-10 13:39:26.256197 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> switch_ivr_play_say.c:1309 Codec Activated > > >> L16 at 8000hz 1 channels 20ms > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> playback(silence_stream://2000) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38_request]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38_request=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38]=[true] > > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completed][200] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.006127 [NOTICE] > > >> mod_dptools.c:1176 Channel > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] has been > > >> answered > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_channel.c:3395 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change RINGING -> ACTIVE > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=sendrecv > > >> a=ptime:20 > > >> a=silenceSupp:off - - - - > > >> a=fmtp:101 0-16 > > >> a=rtpmap:101 telephone-event/8000 > > >> a=rtpmap:0 PCMU/8000 > > >> m=audio 27588 RTP/AVP 0 101 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_sofia.c:856 Local SDP > > >> sofia/external/+14792460547 at flowroute.com > > >> : > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:1046 STUN Success > > >> [72.201.112.131]:[27588] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> answer() > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [api_hangup_hook]=[system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=${last_fax} messages='result: > > >> ${fax_result_text} > > >> sender:${fax_remote_station_id} > > >> pages:${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='${caller_id_name}' > > >> caller_id_number=${caller_id_number}] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\${last_fax} > > >> messages='result: \${fax_result_text} > > >> sender:\${fax_remote_station_id} > > >> pages:\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\${caller_id_name}' > > >> caller_id_number=\${caller_id_number}) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-spymap/+14792460547 > /d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [call_direction]=[local] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(call_direction=local) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action hangup() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> playback(silence_stream://2000) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38_request=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action answer() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\\\${last_fax} > > >> messages='result: \\\${fax_result_text} > > >> sender:\\\${fax_remote_station_id} > > >> pages:\\\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\\\${caller_id_name}' > > >> caller_id_number=\\\${caller_id_number}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test Fax] destination_number(3150) =~ > > >> /^3150$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Test Fax] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain_user] destination_number(3150) =~ > > >> /^\*97$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain_user] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain] destination_number(3150) =~ > > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [send_to_voicemail] > > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->send_to_voicemail] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [extension-intercom] > > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->extension-intercom] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [intercept-ext] destination_number(3150) > > >> =~ /^\*\*(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->intercept-ext] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_return] destination_number(3150) =~ > > >> /^\*69$|^lcr$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_return] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_privacy] destination_number(3150) =~ > > >> /^\*67(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_privacy] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [global] > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > > >> break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->global] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [redial] destination_number(3150) =~ > > >> /^(redial|\*870)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->redial] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [group-intercept] destination_number(3150) > > >> =~ /^\*8$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->group-intercept] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> ANTI-Action > > >> set(call_direction=local) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call-direction] ${call_direction}() =~ > > >> /^(inbound|outbound|local)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call-direction] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [Flowroute.11d] destination_number(3150) > > >> =~ /^\+?(\d{11})$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Flowroute.11d] continue=false > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->3150 in context default > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [NOTICE] > > >> switch_ivr.c:1781 Transfer > > >> sofia/external/+14792460547 at flowroute.com > > >> to > > >> XML[3150 at default] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_ivr.c:1774 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_EXECUTE -> CS_ROUTING > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> transfer(3150 > > >> XML default) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_channel.c:1135 EXPORT (export_vars) > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [outside_call]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(outside_call=true) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> transfer(3150 XML default) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] > > >> destination_number(15203039810) =~ > > >> /^15203039810$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] context(public) =~ /public/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->Test_Number] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_debug] ${call_debug}(false) =~ > > >> /^true$/ break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->call_debug] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(outside_call=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [outside_call] > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->outside_call] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->unloop] continue=false > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->15203039810 in context public > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_channel.c:2003 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change DOWN -> RINGING > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_ROUTING > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:454 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State INIT > > >> going to sleep > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> mod_sofia.c:126 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_INIT -> CS_ROUTING > > >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:454 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_NEW -> CS_INIT > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:5263 Set 2833 dtmf send/recv payload > > >> to 101 > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> switch_core_codec.c:111 > > >> sofia/external/+14792460547 at flowroute.com > > >> Original read > > >> codec set to PCMU:0 > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:3093 Set Codec > > >> sofia/external/+14792460547 at flowroute.com > > >> PCMU/8000 20 > > >> ms 160 samples 64000 bits > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:5134 Audio Codec Compare > > >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > > >> 2013-01-10 13:39:25.806120 [DEBUG] > > >> sofia_glue.c:5134 Audio Codec Compare > > >> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > > >> > > >> a=ptime:20 > > >> a=fmtp:101 0-15 > > >> a=rtpmap:101 telephone-event/8000 > > >> a=fmtp:18 annexb=no > > >> a=rtpmap:18 G729/8000 > > >> m=audio 31632 RTP/AVP 0 18 101 > > >> t=0 0 > > >> c=IN IP4 66.42.121.5 > > >> s=- > > >> o=- 275426305 1357849875 IN IP4 66.42.121.5 > > >> v=0 > > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 > > >> Remote SDP: > > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [received][100] > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_state_machine.c:433 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State NEW > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_NEW > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.786370 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.786370 [NOTICE] > > >> switch_channel.c:968 New Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> > > >> [d276f012-5b65-11e2-beae-8564ff827a25] > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Thu, 10 Jan 2013 09:47:18 -0800 > > >> From: msc at freeswitch.org > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Can you pastebin a console debug log w/ sip trace > > >> of a call failing to negotiate t.38 correctly? > > >> That might help us figure out where to look. > > >> -MC > > >> > > >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig > > >> > >> > wrote: > > >> > > >> Just installed FusionPBX using the script > > >> built for a Centos 6 install. Everything > > >> works great in regards to calls, but my main > > >> goal of this project was to utilize the fax > > >> capabilities of Freeswitch/FusionPBX. I have > > >> tried many things in media bypass, routing to > > >> a fax extension, using an ATA, etc., but I > > >> can't get T38 to negotiate correctly. Can > > >> someone point me in the right direction as to > > >> where I should begin? I am guessing it isn't > > >> supposed to work right out of the box. I am > > >> using Flowroute as my provider right now and > > >> the have confirmed that they utilize T38 in > > >> both directions. Thanks! > > >> > > >> > > >> Robert > > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http:// > lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/4d88b2c5/attachment-0001.html From craig.robert at hotmail.com Sun Jan 20 20:36:01 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Sun, 20 Jan 2013 10:36:01 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: , , , , , , , , , , , , , <50F975E5.2040206@coppice.org>, , , , <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com>, , Message-ID: George, I'm having a hard time understanding your dialplan. I'm not a FS guru so my apologies. It looks like your dialplan is meant for an ATA. I am just trying to get the email portion configured first; specifically inbound. Once I resolve the inbound issues, I'll work on outbound and then move to ATA's. Robert Date: Sun, 20 Jan 2013 09:52:11 -0500 From: gassaad at emassembly.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Here is mine:Inbound:http://pastebin.freeswitch.org/20471Outbound:http://pastebin.freeswitch.org/20470 Thanks,George On Sat, Jan 19, 2013 at 2:54 PM, Robert Craig wrote: OK, I copied everything from the fax extension dial plan into the below pastebin. http://pastebin.freeswitch.org/20466 Robert From: bdfoster at endigotech.com Date: Fri, 18 Jan 2013 12:59:38 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Submit the dialplan you are using to receive the fax. You can use the FreeSWITCH pastebin, http://pastebin.freeswitch.org Sent from my iPhone On Jan 18, 2013, at 12:43 PM, Robert Craig wrote: I believe I'm just using the built in SpanDSP for faxing. I've looked around for 'rxfax' and can't find anything. I did notice that the directory where the faxes were supposed to be stored for the extension I create didn't even exist. I created them, applied the permissions, but it seems like I am missing something because the same error is still there. Robert Date: Fri, 18 Jan 2013 09:13:36 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration If you are using the rxfax app then you specify the file name and location: Make sure that the target directory has read/write permissions. You can use something like "chmod 666 /my/directory" to make it read/write. -MC On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: Where does the file get downloaded to? What should the permissions be? Robert > Date: Sat, 19 Jan 2013 00:18:45 +0800 > From: steveu at coppice.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If the TIFF file cannot be opened on receive you probably have a file > permissions issue (e.g. with the directory in which the file is to be > created). If the TIFF file cannot be opened on transmit there are > various possibilities about the suitability of the file's contents. > > Steve > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > This was an incoming fax. I don't have much control over the other > > end. Why is that error appearing when trying to receive something? > > > > Robert > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > wrote: > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > >> Please try pdf, you should be in business now. > >> Please let us know, > >> > >> On Jan 18, 2013 12:35 AM, "Robert Craig" >> > wrote: > >> > >> OK, so I've made a little bit of progress. The distant end is a > >> Windows Fax server hooked up to an analog line. I created a fax > >> extension in FS, assigned an inbound route to the extension, and > >> made sure the extension had an email address. I believe the fax > >> receive wants to complete, but there is an interesting error in > >> the log about TIF files. Anyone ever seen this before? > >> > >> v=0 > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> c=IN IP4 72.201.112.131 > >> t=0 0 > >> m=image 27714 udptl t38 > >> a=T38FaxVersion:0 > >> a=T38MaxBitRate:9600 > >> a=T38FaxFillBitRemoval > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxUdpEC:t38UDPRedundancy > >> > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > >> ip/port confirmed. > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > >> Negotiation Result > >> ======================================================= > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > >> status off > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > >> country: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > >> vendor: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > >> model: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > >> processing not successful - result (41) TIFF/F file cannot be > >> opened.* > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > >> transferred: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > >> fax pages: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > >> resolution: 0x0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > >> status off > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > >> country: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > >> vendor: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > >> model: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > >> sofia/external/+14792460547 @flowroute.com > >> Restore previous codec PCMU:0. > >> EXECUTE sofia/external/+14792460547 > >> @flowroute.com hangup() > >> > >> Robert > >> > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> OK, I am not extremely concerned with getting an ATA working > >> right now. If I could just get an inbound extension-to-email > >> working it would be great. I'll start from scratch and then let > >> you know what happens. Just to make sure, I am going to create an > >> extension, assign an email address to it, make sure there is an > >> inbound route to that extension, and then test it. > >> > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > >> From: gassaad at emassembly.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> The reason for outbound extension is: > >> When you send a fax from ATA to the outside world, it will dial > >> first to the outbound extension and then after successful receive > >> it will dial out to the intended number. > >> > >> As for the error you have it may be related to your ATA. I am not > >> sure please see why do you get the following message. > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > >> > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > >> > wrote: > >> > >> OK, I've already tried creating one extension for inbound and > >> set an email address. Of course, it failed and the errors are > >> at the bottom of this email. What is the purpose of the > >> outbound extension? > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > >> From: gassaad at emassembly.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Sorry for the delay to get back to you. > >> 1- App --> fax server > >> 2- Create two extensions (one for inbound) another for > >> (outbound) From within either extension you can send faxes > >> "virtual fax" > >> 3- point your create your outbound and inbound dialplans as > >> normal but make it to point to those extensions. > >> > >> Good luck, > >> > >> George > >> > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > >> > > >> wrote: > >> > >> What did you do on the config side to get it to work? > >> I've tried both an Ata and a fax to email extension and > >> they both fail in both directions. > >> > >> Robert > >> > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > >> > > >> wrote: > >> > >> I use Fusion successfully to send / receive fax. > >> The way it works: > >> ATA --->FaxServer. > >> After successful receipt, Fax Server---> DID where > >> you have dialled originally. > >> > >> Cheers, > >> > >> George > >> > >> > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > >> >> > wrote: > >> > >> Did anyone see the issue or what I may have > >> missed in the config? As mentioned earlier, I > >> thought it would after a fresh install. Any help > >> is appreciated. > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Thanks for the quick response. Below is the > >> capture from the log viewer. In this test, I am > >> using a remote fax machine (line from Cox with > >> physical machine) faxing into the "email server" > >> where I created an extension for fax-to-email. > >> Calling number is 1479-246-0547. > >> > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:99 > >> sofia/external/+14792460547 at flowroute.com > >> Standard DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> mod_sofia.c:396 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:559 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State Change > >> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:556 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate Change HANGUP > >> -> DOWN > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1510 Close Channel > >> sofia/external/+14792460547 at flowroute.com > >> [CS_DESTROY] > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1506 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Ended > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1488 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Locked, Waiting on > >> external entities > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:440 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change CS_REPORTING > >> -> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State > >> REPORTING going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:92 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> REPORTING, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:446 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_HANGUP -> CS_REPORTING > >> > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:599 Hangup Command > >> with no Session system(/usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 > >> name=+14792460547-2013-01-10-13-39-26 > >> messages='result: Disconnected after permitted > >> retries sender:ATW-4792460547 pages:0' > >> domain=192.168.4.15 caller_id_name='' > >> caller_id_number=): > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:48 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> HANGUP, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:552 Sending BYE to > >> sofia/external/+14792460547 at flowroute.com > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:503 Channel > >> sofia/external/+14792460547 at flowroute.com > >> hanging up, > >> cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:2687 > >> sofia/external/+14792460547 at flowroute.com > >> skip receive > >> message [APPLICATION_EXEC_COMPLETE] (channel is > >> hungup already) > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:3017 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [KILL] > >> 2013-01-10 13:39:52.216151 [NOTICE] > >> mod_dptools.c:1150 Hangup > >> sofia/external/+14792460547 at flowroute.com > >> [CS_EXECUTE] > >> [NORMAL_CLEARING] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:2994 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change ACTIVE -> HANGUP > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> hangup() > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_codec.c:244 > >> sofia/external/+14792460547 at flowroute.com > >> Restore > >> previous codec PCMU:0. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:516 remote model: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:515 remote vendor: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:514 remote country: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:513 ECM status off > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:509 Total fax pages: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:507 Pages transferred: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > >> Fax Ident > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:505 Remote station id: > >> ATW-4792460547 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:500 Fax processing not > >> successful - result (48) Disconnected after > >> permitted retries. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> 2013-01-10 13:39:38.766190 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:29.576127 [ERR] > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > >> must be between 20 and 10000 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxRateManagement:transferredTCF > >> m=image 31632 udptl t38 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completing][200] > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > >> sofia/external/+14792460547 at flowroute.com > >> Update Callee > >> ID to "+14792460547" <+14792460547> > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [calling][0] > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:2647 Local SDP: > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1379 Raw write codec activation > >> Success L16 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_core_codec.c:219 > >> sofia/external/+14792460547 at flowroute.com > >> Push codec L16:70 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1363 Raw read codec activation > >> Success L16 20000 > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > >> > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_ivr_play_say.c:1682 done playing file > >> silence_stream://2000 > >> 2013-01-10 13:39:26.256197 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> switch_ivr_play_say.c:1309 Codec Activated > >> L16 at 8000hz 1 channels 20ms > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> playback(silence_stream://2000) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38_request]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38_request=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38]=[true] > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completed][200] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.006127 [NOTICE] > >> mod_dptools.c:1176 Channel > >> [sofia/external/+14792460547 at flowroute.com > >> ] has been > >> answered > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_channel.c:3395 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change RINGING -> ACTIVE > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=sendrecv > >> a=ptime:20 > >> a=silenceSupp:off - - - - > >> a=fmtp:101 0-16 > >> a=rtpmap:101 telephone-event/8000 > >> a=rtpmap:0 PCMU/8000 > >> m=audio 27588 RTP/AVP 0 101 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_sofia.c:856 Local SDP > >> sofia/external/+14792460547 at flowroute.com > >> : > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:1046 STUN Success > >> [72.201.112.131]:[27588] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> answer() > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [api_hangup_hook]=[system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=${last_fax} messages='result: > >> ${fax_result_text} > >> sender:${fax_remote_station_id} > >> pages:${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='${caller_id_name}' > >> caller_id_number=${caller_id_number}] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\${last_fax} > >> messages='result: \${fax_result_text} > >> sender:\${fax_remote_station_id} > >> pages:\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\${caller_id_name}' > >> caller_id_number=\${caller_id_number}) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [call_direction]=[local] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(call_direction=local) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action hangup() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> playback(silence_stream://2000) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38_request=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action answer() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\\\${last_fax} > >> messages='result: \\\${fax_result_text} > >> sender:\\\${fax_remote_station_id} > >> pages:\\\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\\\${caller_id_name}' > >> caller_id_number=\\\${caller_id_number}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test Fax] destination_number(3150) =~ > >> /^3150$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Test Fax] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain_user] destination_number(3150) =~ > >> /^\*97$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain_user] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain] destination_number(3150) =~ > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [send_to_voicemail] > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->send_to_voicemail] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [extension-intercom] > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->extension-intercom] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [intercept-ext] destination_number(3150) > >> =~ /^\*\*(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->intercept-ext] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_return] destination_number(3150) =~ > >> /^\*69$|^lcr$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_return] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_privacy] destination_number(3150) =~ > >> /^\*67(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_privacy] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [global] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > >> break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->global] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [redial] destination_number(3150) =~ > >> /^(redial|\*870)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->redial] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [group-intercept] destination_number(3150) > >> =~ /^\*8$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->group-intercept] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> ANTI-Action > >> set(call_direction=local) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call-direction] ${call_direction}() =~ > >> /^(inbound|outbound|local)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call-direction] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [Flowroute.11d] destination_number(3150) > >> =~ /^\+?(\d{11})$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Flowroute.11d] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->3150 in context default > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:25.816280 [NOTICE] > >> switch_ivr.c:1781 Transfer > >> sofia/external/+14792460547 at flowroute.com > >> to > >> XML[3150 at default] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_ivr.c:1774 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_EXECUTE -> CS_ROUTING > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> transfer(3150 > >> XML default) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_channel.c:1135 EXPORT (export_vars) > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [outside_call]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(outside_call=true) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> transfer(3150 XML default) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] > >> destination_number(15203039810) =~ > >> /^15203039810$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] context(public) =~ /public/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->Test_Number] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_debug] ${call_debug}(false) =~ > >> /^true$/ break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->call_debug] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(outside_call=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [outside_call] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->outside_call] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->unloop] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->15203039810 in context public > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_channel.c:2003 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change DOWN -> RINGING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:454 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State INIT > >> going to sleep > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> mod_sofia.c:126 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_INIT -> CS_ROUTING > >> 2013-01-10 13:39:25.806120 [DEBUG] mod_sofia.c:86 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:454 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5831 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_NEW -> CS_INIT > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5263 Set 2833 dtmf send/recv payload > >> to 101 > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> switch_core_codec.c:111 > >> sofia/external/+14792460547 at flowroute.com > >> Original read > >> codec set to PCMU:0 > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:3093 Set Codec > >> sofia/external/+14792460547 at flowroute.com > >> PCMU/8000 20 > >> ms 160 samples 64000 bits > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5134 Audio Codec Compare > >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > >> 2013-01-10 13:39:25.806120 [DEBUG] > >> sofia_glue.c:5134 Audio Codec Compare > >> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > >> > >> a=ptime:20 > >> a=fmtp:101 0-15 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:18 annexb=no > >> a=rtpmap:18 G729/8000 > >> m=audio 31632 RTP/AVP 0 18 101 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849875 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:25.806120 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [received][100] > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_state_machine.c:433 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State NEW > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_NEW > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.786370 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.786370 [NOTICE] > >> switch_channel.c:968 New Channel > >> sofia/external/+14792460547 at flowroute.com > >> > >> [d276f012-5b65-11e2-beae-8564ff827a25] > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Thu, 10 Jan 2013 09:47:18 -0800 > >> From: msc at freeswitch.org > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Can you pastebin a console debug log w/ sip trace > >> of a call failing to negotiate t.38 correctly? > >> That might help us figure out where to look. > >> -MC > >> > >> On Wed, Jan 9, 2013 at 7:42 PM, Robert Craig > >> >> > wrote: > >> > >> Just installed FusionPBX using the script > >> built for a Centos 6 install. Everything > >> works great in regards to calls, but my main > >> goal of this project was to utilize the fax > >> capabilities of Freeswitch/FusionPBX. I have > >> tried many things in media bypass, routing to > >> a fax extension, using an ATA, etc., but I > >> can't get T38 to negotiate correctly. Can > >> someone point me in the right direction as to > >> where I should begin? I am guessing it isn't > >> supposed to work right out of the box. I am > >> using Flowroute as my provider right now and > >> the have confirmed that they utilize T38 in > >> both directions. Thanks! > >> > >> > >> Robert > >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/3258a36e/attachment-0001.html From vbvbrj at gmail.com Sun Jan 20 21:19:58 2013 From: vbvbrj at gmail.com (Mimiko) Date: Sun, 20 Jan 2013 20:19:58 +0200 Subject: [Freeswitch-users] Echo present. Message-ID: <50FC354E.80206@gmail.com> Hello. On my VoIP phone is registered two SIP accounts. One for FS server and one for a VoIP provider. When calling thru FS server I here my echo, while other partie does not hear its echo. When calling thru VoIP's SIP account, I don't here my echo, never other partie hear its echo. So I think that FS server does not remove echo. How to resolve this? -- Mimiko desu. From charlie at charlie.is Sun Jan 20 22:51:18 2013 From: charlie at charlie.is (Charlie Orford) Date: Sun, 20 Jan 2013 20:51:18 +0100 Subject: [Freeswitch-users] mod_voicemail always sends empty emails In-Reply-To: <50FAFA76.3010708@charlie.is> References: <50FAFA76.3010708@charlie.is> Message-ID: <50FC4AB6.6050300@charlie.is> As a follow up: I installed msmtp and changed the mailer-app param in switch.conf.xml to: Then put the following in /etc/msmtprc: account default host mail.domain.com domain fs.domain.com syslog on Now calling an extension and leaving a voicemail doesn't even result in a blank email being sent. It's as if FS never invokes msmtp. There's nothing in /var/log/syslog (where msmtp is configured to log to) and no log of a connection attempt on my mail server. I've spent all day on this trying to get to the bottom of it but am stumped :( Charlie On 19/01/2013 20:56, Charlie Orford wrote: > Hi list, > > Voice mails get correctly saved in > /usr/local/freeswitch/storage/voicemail/default/domain.com/extnum/ but > the email sent by mod_voicemail always arrives blank with the wav file > attachment missing. > > To check the problem wasn't mail server related, I setup a simple > postfix "Local only" installation on the same host as FS. > > Here's the entire contents of a voicemail email sent to postfix by FS > (note: domain.com > is substituted for real domain): > > # cat /var/mail/root > From voicemail at domain.com Sat Jan 19 17:38:03 2013 > Return-Path: > X-Original-To: postmaster at fs.domain.com > Delivered-To: postmaster at fs.domain.com > Received: by fs (Postfix, from userid 1001) > id 9E6AC23E57; Sat, 19 Jan 2013 17:38:03 +0000 (UTC) > Message-Id: <20130119183349.9E6AC23E57 at fs> > Date: Sat, 19 Jan 2013 18:33:49 +0000 (UTC) > From: voicemail at angloeuropean.com > To: undisclosed-recipients:; > > > > And here are the voicemail specific config params for the extension: > > > > > > > > > > > /usr/local/freeswitch/conf/voicemail.tpl exists and looks correct > (contents here: http://pastebin.com/Tr4fvyht ) > > I took a log (via fsi_cli and /log 7) of a call leaving a voicemail to > that extension (here: http://pastebin.com/B1zteGV7 ) but nothing jumps > out. > > Is there anything else I can do to track down what's causing this? > > I'm running the latest stable branch on Debian squeeze: FreeSWITCH > Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 > 2012-12-29 00:17:59Z) > > Thanks, > Charlie > > > > > > > > > From jmesquita at freeswitch.org Sun Jan 20 23:00:10 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 20 Jan 2013 17:00:10 -0300 Subject: [Freeswitch-users] Echo present. In-Reply-To: <50FC354E.80206@gmail.com> References: <50FC354E.80206@gmail.com> Message-ID: Mimiko, FS does not remove echo nor it should. You need to start providing more information on the problems you report on the mailing list otherwise the first response will always be a request for more information. Most probably, if you are hearing echo from your own voice, it must be produced on your endpoint. My best guess would be acoustic echo produced by the handset/headset or if you are using hands-free without decent AEC... What endpoint are you using? Softphone, hard phone or what? JM Jo?o Mesquita FreeSWITCH? Solutions On Sun, Jan 20, 2013 at 4:19 PM, Mimiko wrote: > Hello. > > On my VoIP phone is registered two SIP accounts. One for FS server and > one for a VoIP provider. When calling thru FS server I here my echo, > while other partie does not hear its echo. When calling thru VoIP's SIP > account, I don't here my echo, never other partie hear its echo. So I > think that FS server does not remove echo. How to resolve this? > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/58d200d1/attachment.html From jmesquita at freeswitch.org Sun Jan 20 23:07:50 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 20 Jan 2013 17:07:50 -0300 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: <50F975E5.2040206@coppice.org> <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com> Message-ID: Robert, you clearly still have a permissions problem on this specific folder (according to your dialplan): /usr/local/freeswitch/storage/fax/3150/inbox/ What is the user FreeSWITCH is running under and what are the current permissions on the folder? Jo?o Mesquita FreeSWITCH? Solutions On Sun, Jan 20, 2013 at 3:36 PM, Robert Craig wrote: > George, > > I'm having a hard time understanding your dialplan. I'm not a FS guru > so my apologies. It looks like your dialplan is meant for an ATA. I am just > trying to get the email portion configured first; specifically inbound. > Once I resolve the inbound issues, I'll work on outbound and then move to > ATA's. > > > Robert > > > ------------------------------ > Date: Sun, 20 Jan 2013 09:52:11 -0500 > From: gassaad at emassembly.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Here is mine: > Inbound: > http://pastebin.freeswitch.org/20471 > Outbound: > http://pastebin.freeswitch.org/20470 > > Thanks, > George > > On Sat, Jan 19, 2013 at 2:54 PM, Robert Craig wrote: > > OK, I copied everything from the fax extension dial plan into the below > pastebin. > > http://pastebin.freeswitch.org/20466 > > > Robert > > > ------------------------------ > From: bdfoster at endigotech.com > Date: Fri, 18 Jan 2013 12:59:38 -0500 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Submit the dialplan you are using to receive the fax. You can use the > FreeSWITCH pastebin, http://pastebin.freeswitch.org > > Sent from my iPhone > > On Jan 18, 2013, at 12:43 PM, Robert Craig > wrote: > > I believe I'm just using the built in SpanDSP for faxing. I've looked > around for 'rxfax' and can't find anything. I did notice that the directory > where the faxes were supposed to be stored for the extension I create > didn't even exist. I created them, applied the permissions, but it seems > like I am missing something because the same error is still there. > > > Robert > > > ------------------------------ > Date: Fri, 18 Jan 2013 09:13:36 -0800 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If you are using the rxfax app then you specify the file name and location: > > > > Make sure that the target directory has read/write permissions. You can > use something like "chmod 666 /my/directory" to make it read/write. > > -MC > > On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: > > Where does the file get downloaded to? What should the permissions be? > > Robert > > > > Date: Sat, 19 Jan 2013 00:18:45 +0800 > > From: steveu at coppice.org > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] T38 Configuration > > > > If the TIFF file cannot be opened on receive you probably have a file > > permissions issue (e.g. with the directory in which the file is to be > > created). If the TIFF file cannot be opened on transmit there are > > various possibilities about the suitability of the file's contents. > > > > Steve > > > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > > This was an incoming fax. I don't have much control over the other > > > end. Why is that error appearing when trying to receive something? > > > > > > Robert > > > > > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > > wrote: > > > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > > >> Please try pdf, you should be in business now. > > >> Please let us know, > > >> > > >> On Jan 18, 2013 12:35 AM, "Robert Craig" > >> > wrote: > > >> > > >> OK, so I've made a little bit of progress. The distant end is a > > >> Windows Fax server hooked up to an analog line. I created a fax > > >> extension in FS, assigned an inbound route to the extension, and > > >> made sure the extension had an email address. I believe the fax > > >> receive wants to complete, but there is an interesting error in > > >> the log about TIF files. Anyone ever seen this before? > > >> > > >> v=0 > > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> c=IN IP4 72.201.112.131 > > >> t=0 0 > > >> m=image 27714 udptl t38 > > >> a=T38FaxVersion:0 > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxFillBitRemoval > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > > >> ip/port confirmed. > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > > >> Negotiation Result > > >> ======================================================= > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > > >> status off > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > > >> country: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > > >> vendor: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > > >> model: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > > >> processing not successful - result (41) TIFF/F file cannot be > > >> opened.* > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > > >> transferred: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > > >> fax pages: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > > >> resolution: 0x0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > > >> status off > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > > >> country: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > > >> vendor: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > > >> model: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > > >> sofia/external/+14792460547 @flowroute.com > > >> Restore previous codec PCMU:0. > > >> EXECUTE sofia/external/+14792460547 > > >> @flowroute.com hangup() > > >> > > >> Robert > > >> > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> OK, I am not extremely concerned with getting an ATA working > > >> right now. If I could just get an inbound extension-to-email > > >> working it would be great. I'll start from scratch and then let > > >> you know what happens. Just to make sure, I am going to create an > > >> extension, assign an email address to it, make sure there is an > > >> inbound route to that extension, and then test it. > > >> > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > > >> From: gassaad at emassembly.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> The reason for outbound extension is: > > >> When you send a fax from ATA to the outside world, it will dial > > >> first to the outbound extension and then after successful receive > > >> it will dial out to the intended number. > > >> > > >> As for the error you have it may be related to your ATA. I am not > > >> sure please see why do you get the following message. > > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > > >> > > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > > >> > wrote: > > >> > > >> OK, I've already tried creating one extension for inbound and > > >> set an email address. Of course, it failed and the errors are > > >> at the bottom of this email. What is the purpose of the > > >> outbound extension? > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > > >> From: gassaad at emassembly.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Sorry for the delay to get back to you. > > >> 1- App --> fax server > > >> 2- Create two extensions (one for inbound) another for > > >> (outbound) From within either extension you can send faxes > > >> "virtual fax" > > >> 3- point your create your outbound and inbound dialplans as > > >> normal but make it to point to those extensions. > > >> > > >> Good luck, > > >> > > >> George > > >> > > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > > >> > > > >> wrote: > > >> > > >> What did you do on the config side to get it to work? > > >> I've tried both an Ata and a fax to email extension and > > >> they both fail in both directions. > > >> > > >> Robert > > >> > > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > > >> > > > >> wrote: > > >> > > >> I use Fusion successfully to send / receive fax. > > >> The way it works: > > >> ATA --->FaxServer. > > >> After successful receipt, Fax Server---> DID where > > >> you have dialled originally. > > >> > > >> Cheers, > > >> > > >> George > > >> > > >> > > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > > >> > >> > wrote: > > >> > > >> Did anyone see the issue or what I may have > > >> missed in the config? As mentioned earlier, I > > >> thought it would after a fresh install. Any help > > >> is appreciated. > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Thanks for the quick response. Below is the > > >> capture from the log viewer. In this test, I am > > >> using a remote fax machine (line from Cox with > > >> physical machine) faxing into the "email server" > > >> where I created an extension for fax-to-email. > > >> Calling number is 1479-246-0547. > > >> > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:99 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> mod_sofia.c:396 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:559 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State Change > > >> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:556 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate Change HANGUP > > >> -> DOWN > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1510 Close Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_DESTROY] > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1506 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Ended > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1488 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Locked, Waiting on > > >> external entities > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:440 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change CS_REPORTING > > >> -> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State > > >> REPORTING going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:92 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> REPORTING, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:446 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_HANGUP -> CS_REPORTING > > >> > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:599 Hangup Command > > >> with no Session system(/usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 > > >> name=+14792460547-2013-01-10-13-39-26 > > >> messages='result: Disconnected after permitted > > >> retries sender:ATW-4792460547 pages:0' > > >> domain=192.168.4.15 caller_id_name='' > > >> caller_id_number=): > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:48 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> HANGUP, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:552 Sending BYE to > > >> sofia/external/+14792460547 at flowroute.com > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:503 Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> hanging up, > > >> cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:2687 > > >> sofia/external/+14792460547 at flowroute.com > > >> skip receive > > >> message [APPLICATION_EXEC_COMPLETE] (channel is > > >> hungup already) > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:3017 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [KILL] > > >> 2013-01-10 13:39:52.216151 [NOTICE] > > >> mod_dptools.c:1150 Hangup > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_EXECUTE] > > >> [NORMAL_CLEARING] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:2994 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change ACTIVE -> HANGUP > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> hangup() > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_codec.c:244 > > >> sofia/external/+14792460547 at flowroute.com > > >> Restore > > >> previous codec PCMU:0. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:516 remote model: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:515 remote vendor: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:514 remote country: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:513 ECM status off > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:509 Total fax pages: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:507 Pages transferred: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > > >> Fax Ident > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:505 Remote station id: > > >> ATW-4792460547 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:500 Fax processing not > > >> successful - result (48) Disconnected after > > >> permitted retries. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:38.766190 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:29.576127 [ERR] > > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > > >> must be between 20 and 10000 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxRateManagement:transferredTCF > > >> m=image 31632 udptl t38 > > >> t=0 0 > > >> c=IN IP4 66.42.121.5 > > >> s=- > > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > > >> v=0 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > > >> Remote SDP: > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completing][200] > > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > > >> sofia/external/+14792460547 at flowroute.com > > >> Update Callee > > >> ID to "+14792460547" <+14792460547> > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [calling][0] > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:2647 Local SDP: > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1379 Raw write codec activation > > >> Success L16 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_core_codec.c:219 > > >> sofia/external/+14792460547 at flowroute.com > > >> Push codec L16:70 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1363 Raw read codec activation > > >> Success L16 20000 > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > > >> > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_ivr_play_say.c:1682 done playing file > > >> silence_stream://2000 > > >> 2013-01-10 13:39:26.256197 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> switch_ivr_play_say.c:1309 Codec Activated > > >> L16 at 8000hz 1 channels 20ms > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> playback(silence_stream://2000) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38_request]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38_request=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38]=[true] > > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completed][200] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.006127 [NOTICE] > > >> mod_dptools.c:1176 Channel > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] has been > > >> answered > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_channel.c:3395 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change RINGING -> ACTIVE > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=sendrecv > > >> a=ptime:20 > > >> a=silenceSupp:off - - - - > > >> a=fmtp:101 0-16 > > >> a=rtpmap:101 telephone-event/8000 > > >> a=rtpmap:0 PCMU/8000 > > >> m=audio 27588 RTP/AVP 0 101 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_sofia.c:856 Local SDP > > >> sofia/external/+14792460547 at flowroute.com > > >> : > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:1046 STUN Success > > >> [72.201.112.131]:[27588] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> answer() > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [api_hangup_hook]=[system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=${last_fax} messages='result: > > >> ${fax_result_text} > > >> sender:${fax_remote_station_id} > > >> pages:${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='${caller_id_name}' > > >> caller_id_number=${caller_id_number}] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\${last_fax} > > >> messages='result: \${fax_result_text} > > >> sender:\${fax_remote_station_id} > > >> pages:\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\${caller_id_name}' > > >> caller_id_number=\${caller_id_number}) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-spymap/+14792460547 > /d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [call_direction]=[local] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(call_direction=local) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action hangup() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> playback(silence_stream://2000) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38_request=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action answer() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\\\${last_fax} > > >> messages='result: \\\${fax_result_text} > > >> sender:\\\${fax_remote_station_id} > > >> pages:\\\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\\\${caller_id_name}' > > >> caller_id_number=\\\${caller_id_number}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test Fax] destination_number(3150) =~ > > >> /^3150$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Test Fax] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain_user] destination_number(3150) =~ > > >> /^\*97$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain_user] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain] destination_number(3150) =~ > > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [send_to_voicemail] > > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->send_to_voicemail] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [extension-intercom] > > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->extension-intercom] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [intercept-ext] destination_number(3150) > > >> =~ /^\*\*(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->intercept-ext] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_return] destination_number(3150) =~ > > >> /^\*69$|^lcr$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_return] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_privacy] destination_number(3150) =~ > > >> /^\*67(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_privacy] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [global] > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > > >> break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->global] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [redial] destination_number(3150) =~ > > >> /^(redial|\*870)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->redial] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [group-intercept] destination_number(3150) > > >> =~ /^\*8$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->group-intercept] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> ANTI-Action > > >> set(call_direction=local) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call-direction] ${call_direction}() =~ > > >> /^(inbound|outbound|local)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call-direction] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [Flowroute.11d] destination_number(3150) > > >> =~ /^\+?(\d{11})$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Flowroute.11d] continue=false > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->3150 in context default > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [NOTICE] > > >> switch_ivr.c:1781 Transfer > > >> sofia/external/+14792460547 at flowroute.com > > >> to > > >> XML[3150 at default] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_ivr.c:1774 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_EXECUTE -> CS_ROUTING > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> transfer(3150 > > >> XML default) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_channel.c:1135 EXPORT (export_vars) > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [outside_call]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(outside_call=true) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> transfer(3150 XML default) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] > > >> destination_number(15203039810) =~ > > >> /^15203039810$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] context(public) =~ /public/ > > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->Test_Number] continue=false > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_debug] ${call_debug}(false) =~ > > > >> /^true$/ break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->call_debug] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(outside_call=true) > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [outside_call] > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->outside_call] continue=true > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > > >> Regex > > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->unloop] continue=false > > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->15203039810 in context public > > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> > ... > > [Message clipped] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/992ab48b/attachment-0001.html From jmesquita at freeswitch.org Sun Jan 20 23:13:22 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 20 Jan 2013 17:13:22 -0300 Subject: [Freeswitch-users] video calls, full processing media mode In-Reply-To: <20130120181603.d6c2880b8bf3f7c42f03fded@yandex.ru> References: <20130120181603.d6c2880b8bf3f7c42f03fded@yandex.ru> Message-ID: royj, it is possible (to an extent) but a few questions needs to be answered first. Send us a complete log for a call. On the m line sent from your client, we can't know what video codec you're using since it is using one of the dynamic RTP payload numbers (>=96). Jo?o Mesquita FreeSWITCH? Solutions On Sun, Jan 20, 2013 at 11:16 AM, royj wrote: > > Is there ability to establish video call through FreeSWITCH with full > processing media mode on profile. I see FreeSWITCH answers 183 Session > Progress with 'm=video 0 RTP/AVP 19.', when video-phone offers for example > 'm=video 4002 RTP/AVP 97.' > > -- > Regards, > royj > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/3747a4c1/attachment.html From ptness at gmx.co.uk Sun Jan 20 23:20:21 2013 From: ptness at gmx.co.uk (peter ness) Date: Sun, 20 Jan 2013 21:20:21 +0100 Subject: [Freeswitch-users] bluetooth endpoint Message-ID: <20130120202021.128030@gmx.com> Are they any current soultions that enable bluetooth as an endpoint. From jalsot at gmail.com Mon Jan 21 00:12:09 2013 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Sun, 20 Jan 2013 22:12:09 +0100 Subject: [Freeswitch-users] Remove inband DTMF on record_session Message-ID: Hello, Is there a way to to remove inbad DTMF from the recorded wav file (conditionaly)? If not, how could be that done the easyiest way and a way it could get into FS git? As FS has a great DTMF detector and it has probably all the stuff doing such a feature, I think it would be much better than using an external tool for removing DTMF from finished wav file - and it would consume more I/O probably. Regards, Jalsot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/1e1bfdb5/attachment.html From benoit.raymond at amiconcept.com Mon Jan 21 00:29:04 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Sun, 20 Jan 2013 16:29:04 -0500 Subject: [Freeswitch-users] mod_voicemail always sends empty emails In-Reply-To: <50FC4AB6.6050300@charlie.is> References: <50FAFA76.3010708@charlie.is> <50FC4AB6.6050300@charlie.is> Message-ID: <029601cdf755$2c0be3f0$8423abd0$@amiconcept.com> In conf/autoload_configs/voicemail.conf.xml, the "template-file" param points to a valid template file? By default, it uses voicemail.tpl located in the conf directory. Another thing you can look is by turning debug (fsctl loglevel debug) and deposit a message. In the debug output, you have to see "Sending message to ..." Finally, FS sends email by building an email and store it as a file. It then "cat" the file and pipe the output against the mailer-app like this (I'm taking your msmtp mailer): /bin/cat {email_file} | msmtp -f {vm-mailfrom} {mailer-args} {vm-mailto} Reading a bit about msmtp shows that it requires the -a. You could try adding "-a default" in the mailer-args and see if that will work. Personally, it works just great with sendmail with the following parameters: Hope it helps, Benoit Raymond AMI Concept Inc. Business VoIP Solutions / Web Design Phone: (450) 553-1231 http://www.amiconcept.com -----Message d'origine----- De?: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Charlie Orford Envoy??: 20 janvier 2013 14:51 ??: freeswitch-users at lists.freeswitch.org Objet?: Re: [Freeswitch-users] mod_voicemail always sends empty emails As a follow up: I installed msmtp and changed the mailer-app param in switch.conf.xml to: Then put the following in /etc/msmtprc: account default host mail.domain.com domain fs.domain.com syslog on Now calling an extension and leaving a voicemail doesn't even result in a blank email being sent. It's as if FS never invokes msmtp. There's nothing in /var/log/syslog (where msmtp is configured to log to) and no log of a connection attempt on my mail server. I've spent all day on this trying to get to the bottom of it but am stumped :( Charlie On 19/01/2013 20:56, Charlie Orford wrote: > Hi list, > > Voice mails get correctly saved in > /usr/local/freeswitch/storage/voicemail/default/domain.com/extnum/ but > the email sent by mod_voicemail always arrives blank with the wav file > attachment missing. > > To check the problem wasn't mail server related, I setup a simple > postfix "Local only" installation on the same host as FS. > > Here's the entire contents of a voicemail email sent to postfix by FS > (note: domain.com > is substituted for real domain): > > # cat /var/mail/root > From voicemail at domain.com Sat Jan 19 17:38:03 2013 > Return-Path: > X-Original-To: postmaster at fs.domain.com > Delivered-To: postmaster at fs.domain.com > Received: by fs (Postfix, from userid 1001) > id 9E6AC23E57; Sat, 19 Jan 2013 17:38:03 +0000 (UTC) > Message-Id: <20130119183349.9E6AC23E57 at fs> > Date: Sat, 19 Jan 2013 18:33:49 +0000 (UTC) > From: voicemail at angloeuropean.com > To: undisclosed-recipients:; > > > > And here are the voicemail specific config params for the extension: > > value="true"/> value="postmaster at fs.domain.com"/> > name="vm-email-all-messages" value="true"/> name="vm-message-ext" value="wav"/> name="vm-keep-local-after-email" value="true"/> > > > /usr/local/freeswitch/conf/voicemail.tpl exists and looks correct > (contents here: http://pastebin.com/Tr4fvyht ) > > I took a log (via fsi_cli and /log 7) of a call leaving a voicemail to > that extension (here: http://pastebin.com/B1zteGV7 ) but nothing jumps > out. > > Is there anything else I can do to track down what's causing this? > > I'm running the latest stable branch on Debian squeeze: FreeSWITCH > Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 > 2012-12-29 00:17:59Z) > > Thanks, > Charlie > > > > > > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From charlie at charlie.is Mon Jan 21 00:50:18 2013 From: charlie at charlie.is (Charlie Orford) Date: Sun, 20 Jan 2013 22:50:18 +0100 Subject: [Freeswitch-users] [FIXED] Re: mod_voicemail always sends empty emails In-Reply-To: <50FC4AB6.6050300@charlie.is> References: <50FAFA76.3010708@charlie.is> <50FC4AB6.6050300@charlie.is> Message-ID: <50FC669A.6070602@charlie.is> Posting back in case it helps anyone else. Turns out it was caused by /tmp having spurious permissions (wasn't world writeable). Should have caught it earlier as there was a hint buried in the fs_cli log: 2013-01-19 17:38:02.746424 [WARNING] switch_utils.c:915 Failed to delete file [/tmp/mail.1358617081f8b2] What was strange was that with "sendmail" as the mailer-app, FS still sent an email but with no attachment or body text. With msmtp (or a simple bash wrapper script), nothing appeared to get fired at all. Charlie On 20/01/2013 20:51, Charlie Orford wrote: > As a follow up: > > I installed msmtp and changed the mailer-app param in switch.conf.xml to: > > > > Then put the following in /etc/msmtprc: > > account default > host mail.domain.com > domain fs.domain.com > syslog on > > > Now calling an extension and leaving a voicemail doesn't even result > in a blank email being sent. It's as if FS never invokes msmtp. > There's nothing in /var/log/syslog (where msmtp is configured to log > to) and no log of a connection attempt on my mail server. > > I've spent all day on this trying to get to the bottom of it but am > stumped :( > > Charlie > > > On 19/01/2013 20:56, Charlie Orford wrote: >> Hi list, >> >> Voice mails get correctly saved in >> /usr/local/freeswitch/storage/voicemail/default/domain.com/extnum/ >> but the email sent by mod_voicemail always arrives blank with the wav >> file attachment missing. >> >> To check the problem wasn't mail server related, I setup a simple >> postfix "Local only" installation on the same host as FS. >> >> Here's the entire contents of a voicemail email sent to postfix by FS >> (note: domain.com >> is substituted for real domain): >> >> # cat /var/mail/root >> From voicemail at domain.com Sat Jan 19 17:38:03 2013 >> Return-Path: >> X-Original-To: postmaster at fs.domain.com >> Delivered-To: postmaster at fs.domain.com >> Received: by fs (Postfix, from userid 1001) >> id 9E6AC23E57; Sat, 19 Jan 2013 17:38:03 +0000 (UTC) >> Message-Id: <20130119183349.9E6AC23E57 at fs> >> Date: Sat, 19 Jan 2013 18:33:49 +0000 (UTC) >> From: voicemail at angloeuropean.com >> To: undisclosed-recipients:; >> >> >> >> And here are the voicemail specific config params for the extension: >> >> >> >> >> >> >> >> >> >> >> /usr/local/freeswitch/conf/voicemail.tpl exists and looks correct >> (contents here: http://pastebin.com/Tr4fvyht ) >> >> I took a log (via fsi_cli and /log 7) of a call leaving a voicemail >> to that extension (here: http://pastebin.com/B1zteGV7 ) but nothing >> jumps out. >> >> Is there anything else I can do to track down what's causing this? >> >> I'm running the latest stable branch on Debian squeeze: FreeSWITCH >> Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 >> 2012-12-29 00:17:59Z) >> >> Thanks, >> Charlie >> >> >> >> >> >> >> >> >> > From charlie at charlie.is Mon Jan 21 00:53:05 2013 From: charlie at charlie.is (Charlie Orford) Date: Sun, 20 Jan 2013 22:53:05 +0100 Subject: [Freeswitch-users] mod_voicemail always sends empty emails In-Reply-To: <029601cdf755$2c0be3f0$8423abd0$@amiconcept.com> References: <50FAFA76.3010708@charlie.is> <50FC4AB6.6050300@charlie.is> <029601cdf755$2c0be3f0$8423abd0$@amiconcept.com> Message-ID: <50FC6741.80406@charlie.is> Thanks Benoit. Sadly, it wasn't as exotic as that. Just a case of messed up permissions on /tmp *blush* Charlie On 20/01/2013 22:29, Benoit Raymond wrote: > In conf/autoload_configs/voicemail.conf.xml, the "template-file" param > points to a valid template file? By default, it uses voicemail.tpl located > in the conf directory. > Another thing you can look is by turning debug (fsctl loglevel debug) and > deposit a message. In the debug output, you have to see "Sending message to > ..." > > Finally, FS sends email by building an email and store it as a file. It then > "cat" the file and pipe the output against the mailer-app like this (I'm > taking your msmtp mailer): > > /bin/cat {email_file} | msmtp -f {vm-mailfrom} {mailer-args} {vm-mailto} > > Reading a bit about msmtp shows that it requires the -a. You could try > adding "-a default" in the mailer-args and see if that will work. > > Personally, it works just great with sendmail with the following parameters: > > > > Hope it helps, > > Benoit Raymond > AMI Concept Inc. > Business VoIP Solutions / Web Design > Phone: (450) 553-1231 > http://www.amiconcept.com > > -----Message d'origine----- > De : freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Charlie > Orford > Envoy? : 20 janvier 2013 14:51 > ? : freeswitch-users at lists.freeswitch.org > Objet : Re: [Freeswitch-users] mod_voicemail always sends empty emails > > As a follow up: > > I installed msmtp and changed the mailer-app param in switch.conf.xml to: > > > > Then put the following in /etc/msmtprc: > > account default > host mail.domain.com > domain fs.domain.com > syslog on > > > Now calling an extension and leaving a voicemail doesn't even result in a > blank email being sent. It's as if FS never invokes msmtp. There's nothing > in /var/log/syslog (where msmtp is configured to log to) and no log of a > connection attempt on my mail server. > > I've spent all day on this trying to get to the bottom of it but am stumped > :( > > Charlie > > > On 19/01/2013 20:56, Charlie Orford wrote: >> Hi list, >> >> Voice mails get correctly saved in >> /usr/local/freeswitch/storage/voicemail/default/domain.com/extnum/ but >> the email sent by mod_voicemail always arrives blank with the wav file >> attachment missing. >> >> To check the problem wasn't mail server related, I setup a simple >> postfix "Local only" installation on the same host as FS. >> >> Here's the entire contents of a voicemail email sent to postfix by FS >> (note: domain.com >> is substituted for real domain): >> >> # cat /var/mail/root >> From voicemail at domain.com Sat Jan 19 17:38:03 2013 >> Return-Path: >> X-Original-To: postmaster at fs.domain.com >> Delivered-To: postmaster at fs.domain.com >> Received: by fs (Postfix, from userid 1001) >> id 9E6AC23E57; Sat, 19 Jan 2013 17:38:03 +0000 (UTC) >> Message-Id: <20130119183349.9E6AC23E57 at fs> >> Date: Sat, 19 Jan 2013 18:33:49 +0000 (UTC) >> From: voicemail at angloeuropean.com >> To: undisclosed-recipients:; >> >> >> >> And here are the voicemail specific config params for the extension: >> >> > value="true"/> > value="postmaster at fs.domain.com"/> >> > name="vm-email-all-messages" value="true"/> > name="vm-message-ext" value="wav"/> > name="vm-keep-local-after-email" value="true"/> >> >> >> /usr/local/freeswitch/conf/voicemail.tpl exists and looks correct >> (contents here: http://pastebin.com/Tr4fvyht ) >> >> I took a log (via fsi_cli and /log 7) of a call leaving a voicemail to >> that extension (here: http://pastebin.com/B1zteGV7 ) but nothing jumps >> out. >> >> Is there anything else I can do to track down what's causing this? >> >> I'm running the latest stable branch on Debian squeeze: FreeSWITCH >> Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 >> 2012-12-29 00:17:59Z) >> >> Thanks, >> Charlie >> >> >> >> >> >> >> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dvl36.ripe.nick at gmail.com Mon Jan 21 01:33:15 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Mon, 21 Jan 2013 00:33:15 +0200 Subject: [Freeswitch-users] bluetooth endpoint In-Reply-To: <20130120202021.128030@gmx.com> References: <20130120202021.128030@gmx.com> Message-ID: What does it mean? Bluetooth is wireless protocol. 2013/1/20 peter ness > > Are they any current soultions that enable bluetooth as an endpoint. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/77779c2c/attachment.html From jmesquita at freeswitch.org Mon Jan 21 01:50:41 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 20 Jan 2013 19:50:41 -0300 Subject: [Freeswitch-users] FAX t38_gateway operation confusion (mod_spandsp) In-Reply-To: References: Message-ID: Hello everyone, this is me again. I keep pushing ppl to followup on their stuff so I will put my money where my mouth is. I've made big progress and I will respond to my own questions. Look inline. I am still working on putting the pieces together because there are more scenarios than the ones I anticipated when it comes to t38_gateway. It turns out that FaxA can send a fax to FaxB even tho FaxB was the calling party and therefore, T38 negotiation will be initiated by FSB instead of the expected FSA since it is FaxB emitting CED/Preamble. Anyhow, stay tuned and more will come (probably more questions than answers tho). Regards, Jo?o Mesquita On Thu, Jan 17, 2013 at 12:05 PM, Jo?o Mesquita wrote: > Hello everyone! > > I have been playing around with mod_spandsp's capabilities lately and most > certainly because I am new to the all old faxing stuff (pretty funny, > right?), I have been making a bit of a mess in my head about how a T30->T38 > gateway should work. > > I've set up a testing environment here as follows: > > *Components:* > * > * > 2 analog fax machines (call is FaxA and FaxB) > 2 FXS cards > 2 FreeSWITCH machines (call FSA and FSB) > > *Connection scheme:* > * > * > *FaxA <-> FSA and FaxB <-> FSB* > > Both of these connections are using the respective FXS cards > > *FSA <-SIP-> FSB* > * > * > Both FreeSWITCHes are connected using SIP, of course. > > *Call scenarios:* > > > 1. *FaxA sending a fax to FSB (using rxfax on FSB) - OK* > Dialplan on FSA is: > nocng" /> > data="{fax_t38_enable=true}sofia/internal/FSB" /> > Dialplan on FSB: > > > > > > 2. *FSB sending a a fax to FaxA - OK* > Dialplan on FSA is: > nocng" /> > data="{fax_t38_enable=true}sofia/internal/FSB" /> > Originate line on FSB: > > originate {fax_enable_t38_request=true,fax_enable_t38=true}sofia/internal/FSB 'set:ignore_early_media=true,txfax:/root/fax.tiff' inline > > > 3. *FaxA sending a fax to FaxB in manual mode* > 4. *FaxA sending a fax to FaxB in automatic mode* > 5. *FaxA receiving a fax from FaxB in manual mode* > 6. *FaxA receiving a fax from FaxB in automatic mode* > Here comes the problem. It is not clear to me how this should work > from the dialplan perspective and from the theory as well. Now, as I > understand, t38_gateway works by detecting CED tones as specified on the > code by the line: > *spandsp_fax_detect_session(session, "rw", timeout, > MODEM_CONNECT_TONES_FAX_CED_OR_PREAMBLE, 1, direction, NULL, > t38_gateway_start);* > > > *Questions:* > > If this is true and looking at the T.30 standard, some of the cases > mentioned above make the CED tone optional, is this correct? > I am still not sure if there are cases where CED is optional but V.21 Preamble is not and SpanDSP detects both of them. A question remains... I've read that CED (also called ASN in docs) is the same 2100Hz tone emitted by answering machines. Does that mean we also start fax when there is an answering machine on the other end? > > By talking to Steve Underwood yesterday on IRC, he mentioned that (and I > quote): > "the emitting gateway should detect the signal from the answering FAX > machine and initiate the change to T.38". My understanding is that on 3 for > example, FSA needs to detect CED being sent from FaxB, correct? > That is correct. The reason why it was not working for me is because I've found a bug on latest stable that has been resolved on HEAD (I am doing a bisect tomorrow) that removes the media bug installed by spandsp before it has a chance to detect CED. > > Looking at the T.38 standard Annex D to understand who should initiate a > T.38 re-invite at what instance, I found this phrase: > >> "The receiving gateway should only use the ITU-T V.21 preamble generated >> by the answering FAX endpoint as the trigger for detection, and it should >> generate the INVITE request no more than 200 milliseconds after the ITU-T >> V.21 preamble begins, in order to allow the maximum amount of time for the >> session to transition into ITU-T T.38 mode before the answering FAX >> endpoint begins transmitting NSF, CSI or DIS." > > Googling, I found that the ITU-T V.21 preamble is not CNG or CED tone, but > rather something else... I assume that the _OR_PREAMBLE is exactly that, > correct? > That is also correct. > > I greatly appreciate any help as the Wiki is not very specific on the > analog fax stuff. My intent is to wikify all that and help the folks out > there as well. Fax seems to be mistic for more than myself. > > Regards, > Jo?o Mesquita > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/ac729dca/attachment.html From ptness at gmx.co.uk Mon Jan 21 02:59:55 2013 From: ptness at gmx.co.uk (peter ness) Date: Mon, 21 Jan 2013 00:59:55 +0100 Subject: [Freeswitch-users] bluetooth endpoint Message-ID: <20130120235956.69310@gmx.com> Sorry, I mean connect to a mobile phone endpoint using bluetooth. Similiar to chan_mobile in asterisk. > ----- Original Message ----- > From: Dmitry Lysenko > Sent: 01/20/13 10:33 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] bluetooth endpoint > > What does it mean? Bluetooth is wireless protocol. > > > 2013/1/20 peter ness > > > > > Are they any current soultions that enable bluetooth as an endpoint. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > From jmesquita at freeswitch.org Mon Jan 21 03:12:26 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 20 Jan 2013 21:12:26 -0300 Subject: [Freeswitch-users] mod_xml_cdr and transfers - missing call legs In-Reply-To: References: Message-ID: Cal, I saw from Jira that you were receiving the CDRs for all legs, correct? B legs should always link to their A legs and that's how you find what happened and when. Were you able to? Jo?o Mesquita FreeSWITCH? Solutions On Fri, Jan 18, 2013 at 8:18 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello, > > So, we are currently testing mod_xml_cdr and having some difficulties > getting all the correct data. > > I actually have a fairly well commented dump of this happening (albeit > from an unrelated bug); > > http://jira.freeswitch.org/secure/attachment/17781/debug-18-01-2013.txt > > Take the following scenario: > > From user/2000 TO user/2001 > from user/2001 to user/2002 via user/2000 > from user/2002 to gw/447534971120 via user/2001 > from gw/447534971120 to ivr/2003 via user/2001 > from gw/447534971120 to gw/442477062086 via ivr/2003 > HANGUP from gw/447534971120 > > At this point the CDRs are posted back to our web app (both A and B leg).. > however, it doesn't seem to contain all the above call events.. > > The end goal for me was to have each stage of a call logged and linked.. > so if a call is transferred 10 times, I will see 10 CDR legs. > > So, my questions are; > > * Is this behaviour expected? > * Am I approaching this from the wrong angle? > * Are there better ways to collect CDRs? > > Any and all feedback would be much appreciated. > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/f291d82b/attachment-0001.html From jmesquita at freeswitch.org Mon Jan 21 03:41:40 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 20 Jan 2013 21:41:40 -0300 Subject: [Freeswitch-users] mod_xml_curl - request for comments In-Reply-To: References: <804D48104511D4468F0D60DF9D3100350B0A1102@MAIL.millicorp.com> Message-ID: Cal, I filled in a couple of TODO you had there. Take a look. On the directory, it might be best to separate requests as initiated by module so that they are more clear. What do you think? Take a look at what I did and see if it is clear to you. Jo?o Mesquita FreeSWITCH? Solutions On Fri, Jan 18, 2013 at 8:10 PM, Michael Collins wrote: > Anything that is globally applicable probably shouldn't be documented > solely on this page. It would be better to codify that in a separate page > that can be linked-to from various other pages. I would cut & paste that > information into a diff wiki page and link to it from the xml curl page and > see what that looks like. > > -MC > > > On Fri, Jan 18, 2013 at 9:34 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> I agree that having some alternative input/ideas on this page would >> certainly be a good thing. >> >> I'm almost tempted to say that this page doesn't need to explain/detail >> any of the channel variable contexts (the ones where it explains Applicable >> common fields etc), as this is really FreeSWITCH wide, >> and interpretation of these variables mostly depends on your own business >> logic. >> >> Removing this info would make the page a lot more read-able, but at the >> same time, might make it more difficult to understand for new users. >> >> Perhaps a warning at the top somewhere explaining this might be more >> suitable? >> >> Cal >> >> On Fri, Jan 18, 2013 at 4:49 PM, Michael Collins wrote: >> >>> Agreed, this is a significant improvement. Keep plugging away and if >>> anyone else has the time and knowledge to add to this page that would be >>> fantastic. >>> -MC >>> >>> On Fri, Jan 18, 2013 at 6:50 AM, Tim Meade wrote: >>> >>>> Hi Cal?.**** >>>> >>>> ** ** >>>> >>>> I think it looks MUCH better. **** >>>> >>>> ** ** >>>> >>>> Great job,**** >>>> >>>> ** ** >>>> >>>> Tim**** >>>> >>>> ** ** >>>> >>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal >>>> Leeming [Simplicity Media Ltd] >>>> *Sent:* Friday, January 18, 2013 9:33 AM >>>> *To:* FreeSWITCH Users Help >>>> *Subject:* [Freeswitch-users] mod_xml_curl - request for comments**** >>>> >>>> ** ** >>>> >>>> Hello,**** >>>> >>>> ** ** >>>> >>>> I've been re-reading my changes to mod_xml_curl documentation, and I'm >>>> still not entirely happy with it.**** >>>> >>>> ** ** >>>> >>>> I can't tell if it genuinely needs a different approach, or if I'm just >>>> being fussy.**** >>>> >>>> ** ** >>>> >>>> Therefore, constructive criticism is very much welcome - even if it >>>> means the whole damn thing has to be re-done again.**** >>>> >>>> ** ** >>>> >>>> http://wiki.freeswitch.org/wiki/Mod_xml_curl**** >>>> >>>> ** ** >>>> >>>> Thanks**** >>>> >>>> >>>> Cal**** >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/96a566c0/attachment.html From support at ecn.net.au Mon Jan 21 03:47:05 2013 From: support at ecn.net.au (support at ecn.net.au) Date: Mon, 21 Jan 2013 10:47:05 +1000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> <20130118102614.GA27116@eagle.cupis.co.uk> <8A80C99638A5E24E83696EAE33A9B6B968B5302824@twbexmbx001.ecn.hosting> , Message-ID: <8A80C99638A5E24E83696EAE33A9B6B968B5302828@twbexmbx001.ecn.hosting> Hi Richard many thanks for your message. We're having some odd issues that is for sure (and it is likely us no doubt). Tried the settings you suggested (strip down FS and just proxy the media) - however we still can't get the PBX's on the inside of the SBC to detect inband, very strange - if we record the call or tcpdump the RTP and play it back in wireshark the DTMF tones are present. Moving forward from that, I am happy to use start_dtmf to convert inband->rfc2833, however we have a major problem with start_dtmf. The issue is that in FS 1.2/1.3 (and NOT in FS 1.0.6) when we start_dtmf it causes the auto to break up on some message playbacks from the internal PBX's. We've wiresharked both the A and B leg and I can confirm that the audio on the B leg (PBX ->FS SBC) is perfect, but the audio on the A-Leg (FS-SBC -> TELCO/SIPPROVIDER) has significant breakups. If you browse www.ecn.net.au/rtp_stream_pbx_to_fs.jpg (the rtp stream from wireshark from PBX->FS) www.ecn.net.au/rtp_stream_fs_to_telco.jpg (the rtp stream from FS->TELCO). On the stream from FS to the Telco the arrows show sections of the rtp where there is no audio, however it's getting to the FS just that FS seems to be dropping audio. (we are running G711). This only happens with using start_dtmf before bridging the call and when we are running FS 1.2/1.3 (on 1.0.6 the start_dtmf does not exhibit this issue). The stream is an audio stream from an IVR on an Asterisk platform. Very odd! Any ideas much appreciated. What we don't get is why 1.0.6 doesn't show the problem, where as all 1.2's we've tried do (change of functionality or are we missing some configuration directives?) Best Regards Mark ________________________________________ From: Richard Brady [rnbrady at gmail.com] Sent: Sunday, 20 January 2013 9:08 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SBC In-band DTMF Hey Mark A few thoughts: 1. If you are using start_dtmf on the FS box then you are making it do media processing which is fine but this is a change from what the OpenSER / RTPproxy were doing for you. If you want it to be a drop-in replacement you might want to look at some of the following: proxy_media inbound-proxy-media http://wiki.freeswitch.org/wiki/Proxy_Media disable-transcoding codec_string / ep_codec_string inherit_codec http://wiki.freeswitch.org/wiki/Codec_negotiation 2. It's dtmf-type in the profile or dtmf_type in the dialplan. Not dtmf_mode as you mention in an earlier message. 3. Your post mentioning PT 18 and the annexb parameter suggests you are using G729, which does not support inband DTMF reliably, if at all. 4. You will need to put the sip_append_audio_sdp before the pre_answer (which you should get rid off as suggested already). 5. if you do want FS to convert from inband to rfc2833, there was a previous post suggesting spandsp_start_dtmf may perform better than start_dtmf (although I can't verify this). Regards, Richard On 18 January 2013 16:12, Steven Ayre > wrote: Did you try removing the rfc2833-pt param and setting on the outgoing sofia profile to the PBX? Inband tones will always be passed over, FS would actively need to detect the tone and remove it from the media to avoid doing so. So the PBX is not looking inband for it. Since there's no way in SIP to notify that inband tones are in use the PBX is probably checking for out-of-band DTMF and not checking inband at all if out-of-band has been negotiated - so what you probably need to is disable out of band media entirely. As far as comparing FS behaviour to OpenSER goes I would avoid it. They're very different products. Understand the difference between a SIP proxy (OpenSER) and a B2BUA (FreeSWITCH). OpenSER only forwards SIP messages with media terminating on the endpoints. FS negotiates 2 separate calls including media and terminates both calls on FS and passes media between them. As such OpenSER doesn't do any media negotiation and FS does which will make their behaviour rather different. -Steve On 18 January 2013 15:20, support at ecn.net.au > wrote: Yes, we read that and tried to set that when attempting multiple configurations (attempting to pass inband dtmf). Currently we can't get Freeswitch in an SBC role to pass the inband in the media stream at all (see earlier email) - also having odd issues with start_dtmf when accessing Asterisk IVR's (again very odd). the only difference we can see (between FS and our legacy openser) is the Invite pushed on the bleg on Openser included the fmtp:18 annexb=no directive (as per the telco's SDP to us); however on FS it doesn't forward these (and we can't seem to force FS to add these attributes to the SDP on the bleg). Again the start_dtmf issues exists only in 1.2 and 1.3 FS , in the 1.0.6 the issue is not present (is this a bug possibly?) Any ideas? Kind Regards, Mark ________________________________________ From: Paul Cupis [paul at cupis.co.uk] Sent: Friday, 18 January 2013 8:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SBC In-band DTMF On Fri, Jan 18, 2013 at 01:57:36AM +0000, Steven Ayre wrote: > > > What're you hoping to do here? This sets the payload type value. 0 is a > valid pt. This won't disable it, it'll instead use the value 0. That > happens to be the one for G711, so if that codec is enabled you're just > going to get a conflict. If the rfc2833-pt is set to a value <96 it is outside the allowed range for dynamic payload negotiation and FS will not offer RFC2833 support in the SDP. ref: sofia_glue.c Regards, _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gvvsubhashkumar at gmail.com Mon Jan 21 04:41:18 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Mon, 21 Jan 2013 07:11:18 +0530 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: Michael, Did you see any problem in our Freeswitch configuration after you gone through the logs. Thanks, Subhash. On Fri, Jan 18, 2013 at 11:57 AM, Subhash wrote: > Yes,we are trying to bridge PSTN and VOIP call. > > We posted the log in pastebin.Please use the below URL to access the > freeswitch log. > > http://pastebin.freeswitch.org/20461 > > > Thanks, > Subhash. > > > On Thu, Jan 17, 2013 at 10:25 PM, Moises Silva wrote: > >> On Thu, Jan 17, 2013 at 4:41 AM, Subhash wrote: >> >>> Hi Michael, >>> >>> Below is the snippet of the freeswitch log. >>> >>> Success Call : >>> >>> 2013-01-16 00:58:58.314163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 >>> 2013-01-16 00:58:58.314163 [DEBUG] ftdm_io.c:3038 [s4c1][4:1] Created >>> DTMF buffer >>> 2013-01-16 00:58:58.314163 [DEBUG] switch_ivr_bridge.c:393 Send signal >>> FreeTDM/4:1/2001 [BREAK] >>> 2013-01-16 00:58:58.314163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >>> DTMF [9] >>> 2013-01-16 00:59:06.414163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 1:640 >>> 2013-01-16 00:59:06.414163 [DEBUG] switch_ivr_bridge.c:393 Send signal >>> FreeTDM/4:1/2001 [BREAK] >>> 2013-01-16 00:59:06.414163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >>> DTMF [1] >>> 2013-01-16 00:59:06.774163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 >>> 2013-01-16 00:59:06.774163 [DEBUG] switch_ivr_bridge.c:393 Send signal >>> FreeTDM/4:1/2001 [BREAK] >>> 2013-01-16 00:59:06.774163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >>> DTMF [9] >>> 2013-01-16 00:59:09.214164 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read >>> wanpipe event 3 >>> 2013-01-16 00:59:09.254162 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read >>> wanpipe event 3 >>> 2013-01-16 00:59:09.254162 [DEBUG] ftmod_wanpipe.c:1547 [s4c1][4:1] >>> Queuing wanpipe DTMF: 7 >>> 2013-01-16 00:59:09.254162 [DEBUG] ftdm_io.c:3783 [s4c1][4:1] Queuing >>> DTMF 7 (debug = 0) >>> 2013-01-16 00:59:09.254162 [DEBUG] mod_freetdm.c:811 Queuing DTMF [7] in >>> channel FreeTDM/4:1/2001 device 4:1 >>> 2013-01-16 00:59:09.254162 [DEBUG] switch_ivr_bridge.c:393 Send signal >>> sofia/VOIPBRD1/2001@**.**.**.** [BREAK] >>> 2013-01-16 00:59:09.514163 [DEBUG] switch_rtp.c:2589 Queue digit delay >>> of 40ms >>> 2013-01-16 00:59:09.594164 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read >>> wanpipe event 3 >>> 2013-01-16 00:59:09.654166 [DEBUG] ftmod_wanpipe.c:1678 [s4c1][4:1] read >>> wanpipe event 3 >>> 2013-01-16 00:59:09.654166 [DEBUG] ftmod_wanpipe.c:1547 [s4c1][4:1] >>> Queuing wanpipe DTMF: 1 >>> 2013-01-16 00:59:09.654166 [DEBUG] ftdm_io.c:3783 [s4c1][4:1] Queuing >>> DTMF 1 (debug = 0) >>> 2013-01-16 00:59:09.674165 [DEBUG] mod_freetdm.c:811 Queuing DTMF [1] in >>> channel FreeTDM/4:1/2001 device 4:1 >>> 2013-01-16 00:59:09.674165 [DEBUG] switch_ivr_bridge.c:393 Send signal >>> sofia/VOIPBRD1/2001@**.**.**.** [BREAK] >>> 013-01-16 00:59:09.934163 [DEBUG] switch_rtp.c:2589 Queue digit delay of >>> 40ms >>> 2013-01-16 00:59:11.234163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 >>> 2013-01-16 00:59:11.234163 [DEBUG] switch_ivr_bridge.c:393 Send signal >>> FreeTDM/4:1/2001 [BREAK] >>> 2013-01-16 00:59:11.234163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >>> DTMF [9] >>> 2013-01-16 00:59:11.394163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 >>> 2013-01-16 00:59:11.394163 [DEBUG] switch_ivr_bridge.c:393 Send signal >>> FreeTDM/4:1/2001 [BREAK] >>> 2013-01-16 00:59:11.394163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >>> DTMF [9] >>> Failed DTMF call : >>> >>> 2013-01-16 01:02:27.694163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9:640 >>> 2013-01-16 01:02:27.694163 [DEBUG] switch_ivr_bridge.c:393 Send signal >>> FreeTDM/4:1/2001 [BREAK] >>> 2013-01-16 01:02:27.694163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >>> DTMF [9] >>> 2013-01-16 01:02:32.834164 [INFO] ftmod_sangoma_isdn_stack_rcv.c:232 >>> [s4c1][4:1] Received DISCONNECT (suId:1 suInstId:2 spInstId:2) >>> 2013-01-16 01:02:32.834164 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:543 >>> [s4c1][4:1] Processing DISCONNECT (suId:1 suInstId:2 >>> >>> >> This looks like incoming DTMF from the RTP leg going out in the TDM leg? >> if so, the problem would be why switch_rtp.c is "detecting" received DTMF >> when there is none? >> >> Please post in pastebin the full debug log (from call start, bridge, to >> hangup etc) for one success call followed by the call with the dtmf problem. >> >> >> *Moises Silva >> **Manager, Software Engineering*** >> >> msilva at sangoma.com >> >> Sangoma Technologies >> >> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >> >> >> t. +1 800 388 2475 (N. America) >> >> t. +1 905 474 1990 x128 >> >> f. +1 905 474 9223 >> >> >> >> ** >> >> Products >> | Solutions >> | Events >> | Contact >> | Wiki >> | Facebook >> | Twitter`| >> | YouTube >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/95e315be/attachment.html From moises.silva at gmail.com Mon Jan 21 04:42:03 2013 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 20 Jan 2013 20:42:03 -0500 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: On Fri, Jan 18, 2013 at 1:27 AM, Subhash wrote: > Yes,we are trying to bridge PSTN and VOIP call. > > We posted the log in pastebin.Please use the below URL to access the > freeswitch log. > > http://pastebin.freeswitch.org/20461 > > 1. 013-01-16 01:02:26.954165 [INFO] ftmod_sangoma_isdn_stack_rcv.c:144 [ s4c1][4:1] Received CONNECT/CONNECT ACK (suId:1 suInstId:2 spInstId:2 ces:0) 2. 2013-01-16 01:02:26.954165 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:285 [s4c1][4:1] Processing CONNECT/CONNECT ACK (suId:1 suInstId:2 spInstId:2 ces:0) 3. 2013-01-16 01:02:26.974165 [DEBUG] switch_rtp.c:3596 Correct ip/port confirmed. 4. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:1002 [s4c1][4:1] First packet read stats: Rx queue len: 4, Rx queue size: 10 5. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:938 [s4c1][4:1] First packet write stats: Tx queue len: 1, Tx queue size: 5, Tx idle: 10 6. 2013-01-16 01:02:27.694163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9: 640 7. 2013-01-16 01:02:27.694163 [DEBUG] switch_ivr_bridge.c:393 Send signal FreeTDM/4:1/2001 [BREAK] 8. 2013-01-16 01:02:27.694163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating DTMF [9] The DTMF comes from the RTP side, not FreeTDM (as shown by the log above). You have to verify using tcpdump/wireshark if there is any real DTMF being received. I'd do exactly the same test again, but this time I would also start a wireshark trace. Then please pastebin the trace again and upload the pcap trace to cloudshark (http://www.cloudshark.org/) and paste the link here. Moy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/7f0041e4/attachment-0001.html From moises.silva at gmail.com Mon Jan 21 04:48:16 2013 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 20 Jan 2013 20:48:16 -0500 Subject: [Freeswitch-users] Remove inband DTMF on record_session In-Reply-To: References: Message-ID: On Sun, Jan 20, 2013 at 4:12 PM, Tamas Jalsovszky wrote: > Hello, > > Is there a way to to remove inbad DTMF from the recorded wav file > (conditionaly)? > If not, how could be that done the easyiest way and a way it could get > into FS git? > > As FS has a great DTMF detector and it has probably all the stuff doing > such a feature, I think it would be much better than using an external tool > for removing DTMF from finished wav file - and it would consume more I/O > probably. > > Removing it would not be that hard (ie, inserting silence instead). Removing it without leaving a trace any DTMF was ever there sounds a tad more difficult. If you're ok with inserting a silence instead of the tone, extending the record_session app to detect and remove DTMF is probably what I'd do. It is also possible to create a new app that monitors the audio stream and silence the stream when DTMF is detected, but since the detection takes a few milliseconds (~50ms), there would be a bit of "bleeding" of DTMF in the signal, unless the new app also delays the signal by 50ms and then has a way to strip the DTMF completely. *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130120/d88fc415/attachment.html From rnbrady at gmail.com Mon Jan 21 05:34:06 2013 From: rnbrady at gmail.com (Richard Brady) Date: Mon, 21 Jan 2013 02:34:06 +0000 Subject: [Freeswitch-users] SBC In-band DTMF In-Reply-To: <8A80C99638A5E24E83696EAE33A9B6B968B5302828@twbexmbx001.ecn.hosting> References: <8A80C99638A5E24E83696EAE33A9B6B968B53027E7@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B53027FC@twbexmbx001.ecn.hosting> <20130118102614.GA27116@eagle.cupis.co.uk> <8A80C99638A5E24E83696EAE33A9B6B968B5302824@twbexmbx001.ecn.hosting> <8A80C99638A5E24E83696EAE33A9B6B968B5302828@twbexmbx001.ecn.hosting> Message-ID: > Tried the settings you suggested (strip down FS and just proxy the media) > - however we still can't get the PBX's on the inside of the SBC to detect > inband, very strange - if we record the call or tcpdump the RTP and play it > back in wireshark the DTMF tones are present. > The fact you can't get this working means something strange is happening. Personally I wouldn't trust this environment enough to start playing with things like DTMF transcoding. Did you verify in the capture that you are not advertising RFC2833 to the PBX? Very hard to diagnose without full traces but appreciate you can't share these on the interwebs. Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/e7b1e167/attachment.html From gvvsubhashkumar at gmail.com Mon Jan 21 05:43:31 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Mon, 21 Jan 2013 08:13:31 +0530 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: But we don't see any problem when both endpoints are SIP. Thanks, Subhash. On Mon, Jan 21, 2013 at 7:12 AM, Moises Silva wrote: > On Fri, Jan 18, 2013 at 1:27 AM, Subhash wrote: > >> Yes,we are trying to bridge PSTN and VOIP call. >> >> We posted the log in pastebin.Please use the below URL to access the >> freeswitch log. >> >> http://pastebin.freeswitch.org/20461 >> >> > > > 1. 013-01-16 01:02:26.954165 [INFO] ftmod_sangoma_isdn_stack_rcv.c:144 > [s4c1][4:1] Received CONNECT/CONNECT ACK (suId:1 suInstId:2 spInstId:2 > ces:0) > 2. 2013-01-16 01:02:26.954165 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c: > 285 [s4c1][4:1] Processing CONNECT/CONNECT ACK (suId:1 suInstId:2 > spInstId:2 ces:0) > 3. 2013-01-16 01:02:26.974165 [DEBUG] switch_rtp.c:3596 Correct > ip/port confirmed. > 4. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:1002 [s4c1][4:1] First > packet read stats: Rx queue len: 4, Rx queue size: 10 > 5. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:938 [s4c1][4:1] First > packet write stats: Tx queue len: 1, Tx queue size: 5, Tx idle: 10 > 6. 2013-01-16 01:02:27.694163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF 9 > :640 > 7. 2013-01-16 01:02:27.694163 [DEBUG] switch_ivr_bridge.c:393 Send > signal FreeTDM/4:1/2001 [BREAK] > 8. 2013-01-16 01:02:27.694163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating > DTMF [9] > > > The DTMF comes from the RTP side, not FreeTDM (as shown by the log above). > You have to verify using tcpdump/wireshark if there is any real DTMF being > received. > > I'd do exactly the same test again, but this time I would also start a > wireshark trace. Then please pastebin the trace again and upload the pcap > trace to cloudshark (http://www.cloudshark.org/) and paste the link here. > > Moy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/f1bd6685/attachment.html From mike at jerris.com Mon Jan 21 06:23:04 2013 From: mike at jerris.com (Michael Jerris) Date: Sun, 20 Jan 2013 22:23:04 -0500 Subject: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 In-Reply-To: <50FA4D8D.4000405@gmail.com> References: <50F909AD.2010308@gmail.com> <50F91490.7080006@gmail.com> <50F9A0C0.8050401@gmail.com> <1C7BD7EC-8865-40EE-A94D-5D2D785B99E4@gmail.com> <50FA4D8D.4000405@gmail.com> Message-ID: <2969BAE1-3B46-445F-B867-B6562E9BD583@jerris.com> So correct permissions so it can write the core file? On Jan 19, 2013, at 2:38 AM, Mimiko wrote: > On 19.01.2013 00:10, Steven Ayre wrote: >> They should go to the cwd, get the FS process and find it by 'ls -ld /proc/PID/cwd' >> >> Also check the ulimit settings, -c limits the size of coredumps. >> >> Steve on iPhone > > Yep, I know the directory where FS is running. But FS user does not have > write permission to that directory. "ulimit" is also set as per wiki. > > -- > Mimiko desu. From dvl36.ripe.nick at gmail.com Mon Jan 21 08:01:59 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Mon, 21 Jan 2013 07:01:59 +0200 Subject: [Freeswitch-users] bluetooth endpoint In-Reply-To: <20130120235956.69310@gmx.com> References: <20130120235956.69310@gmx.com> Message-ID: Only gsmopen exist in freeswitch at this moment for this purposes, but it does not support bluetooth link with cell phones. 2013/1/21 peter ness > Sorry, I mean connect to a mobile phone endpoint using bluetooth. Similiar > to chan_mobile in asterisk. > > > ----- Original Message ----- > > From: Dmitry Lysenko > > Sent: 01/20/13 10:33 PM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] bluetooth endpoint > > > > What does it mean? Bluetooth is wireless protocol. > > > > > > 2013/1/20 peter ness > > > > > > > > Are they any current soultions that enable bluetooth as an endpoint. > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/64b1d26b/attachment.html From craig.robert at hotmail.com Mon Jan 21 08:59:32 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Sun, 20 Jan 2013 22:59:32 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: , , , , , , , , , , , , , <50F975E5.2040206@coppice.org>, , , , <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com>, , , Message-ID: As far as I know, FS is running as root. Below are the permissions. <13>:ls -ld storage/fax/3150/inbox drw-rw-rw- 2 root root 4096 Jan 18 10:28 storage/fax/3150/inbox I've gone from the 'freeswitch' folder all the way into the inbox folder verifying the same permissions down the line. Should it include 'execute' permissions as well? Whats even more interesting is that when I create a new fax extension, it doesn't create the folder, such as 3150. I have to manually create it. Robert Date: Sun, 20 Jan 2013 17:07:50 -0300 From: jmesquita at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Robert, you clearly still have a permissions problem on this specific folder (according to your dialplan):/usr/local/freeswitch/storage/fax/3150/inbox/ What is the user FreeSWITCH is running under and what are the current permissions on the folder? Jo?o Mesquita FreeSWITCH? Solutions On Sun, Jan 20, 2013 at 3:36 PM, Robert Craig wrote: George, I'm having a hard time understanding your dialplan. I'm not a FS guru so my apologies. It looks like your dialplan is meant for an ATA. I am just trying to get the email portion configured first; specifically inbound. Once I resolve the inbound issues, I'll work on outbound and then move to ATA's. Robert Date: Sun, 20 Jan 2013 09:52:11 -0500 From: gassaad at emassembly.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Here is mine: Inbound:http://pastebin.freeswitch.org/20471Outbound:http://pastebin.freeswitch.org/20470 Thanks,George On Sat, Jan 19, 2013 at 2:54 PM, Robert Craig wrote: OK, I copied everything from the fax extension dial plan into the below pastebin. http://pastebin.freeswitch.org/20466 Robert From: bdfoster at endigotech.com Date: Fri, 18 Jan 2013 12:59:38 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Submit the dialplan you are using to receive the fax. You can use the FreeSWITCH pastebin, http://pastebin.freeswitch.org Sent from my iPhone On Jan 18, 2013, at 12:43 PM, Robert Craig wrote: I believe I'm just using the built in SpanDSP for faxing. I've looked around for 'rxfax' and can't find anything. I did notice that the directory where the faxes were supposed to be stored for the extension I create didn't even exist. I created them, applied the permissions, but it seems like I am missing something because the same error is still there. Robert Date: Fri, 18 Jan 2013 09:13:36 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration If you are using the rxfax app then you specify the file name and location: Make sure that the target directory has read/write permissions. You can use something like "chmod 666 /my/directory" to make it read/write. -MC On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: Where does the file get downloaded to? What should the permissions be? Robert > Date: Sat, 19 Jan 2013 00:18:45 +0800 > From: steveu at coppice.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If the TIFF file cannot be opened on receive you probably have a file > permissions issue (e.g. with the directory in which the file is to be > created). If the TIFF file cannot be opened on transmit there are > various possibilities about the suitability of the file's contents. > > Steve > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > This was an incoming fax. I don't have much control over the other > > end. Why is that error appearing when trying to receive something? > > > > Robert > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > wrote: > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > >> Please try pdf, you should be in business now. > >> Please let us know, > >> > >> On Jan 18, 2013 12:35 AM, "Robert Craig" >> > wrote: > >> > >> OK, so I've made a little bit of progress. The distant end is a > >> Windows Fax server hooked up to an analog line. I created a fax > >> extension in FS, assigned an inbound route to the extension, and > >> made sure the extension had an email address. I believe the fax > >> receive wants to complete, but there is an interesting error in > >> the log about TIF files. Anyone ever seen this before? > >> > >> v=0 > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> c=IN IP4 72.201.112.131 > >> t=0 0 > >> m=image 27714 udptl t38 > >> a=T38FaxVersion:0 > >> a=T38MaxBitRate:9600 > >> a=T38FaxFillBitRemoval > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxUdpEC:t38UDPRedundancy > >> > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > >> ip/port confirmed. > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > >> Negotiation Result > >> ======================================================= > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > >> status off > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > >> country: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > >> vendor: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > >> model: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > >> processing not successful - result (41) TIFF/F file cannot be > >> opened.* > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > >> transferred: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > >> fax pages: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > >> resolution: 0x0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > >> status off > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > >> country: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > >> vendor: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > >> model: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > >> sofia/external/+14792460547 @flowroute.com > >> Restore previous codec PCMU:0. > >> EXECUTE sofia/external/+14792460547 > >> @flowroute.com hangup() > >> > >> Robert > >> > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> OK, I am not extremely concerned with getting an ATA working > >> right now. If I could just get an inbound extension-to-email > >> working it would be great. I'll start from scratch and then let > >> you know what happens. Just to make sure, I am going to create an > >> extension, assign an email address to it, make sure there is an > >> inbound route to that extension, and then test it. > >> > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > >> From: gassaad at emassembly.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> The reason for outbound extension is: > >> When you send a fax from ATA to the outside world, it will dial > >> first to the outbound extension and then after successful receive > >> it will dial out to the intended number. > >> > >> As for the error you have it may be related to your ATA. I am not > >> sure please see why do you get the following message. > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > >> > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > >> > wrote: > >> > >> OK, I've already tried creating one extension for inbound and > >> set an email address. Of course, it failed and the errors are > >> at the bottom of this email. What is the purpose of the > >> outbound extension? > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > >> From: gassaad at emassembly.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Sorry for the delay to get back to you. > >> 1- App --> fax server > >> 2- Create two extensions (one for inbound) another for > >> (outbound) From within either extension you can send faxes > >> "virtual fax" > >> 3- point your create your outbound and inbound dialplans as > >> normal but make it to point to those extensions. > >> > >> Good luck, > >> > >> George > >> > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > >> > > >> wrote: > >> > >> What did you do on the config side to get it to work? > >> I've tried both an Ata and a fax to email extension and > >> they both fail in both directions. > >> > >> Robert > >> > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > >> > > >> wrote: > >> > >> I use Fusion successfully to send / receive fax. > >> The way it works: > >> ATA --->FaxServer. > >> After successful receipt, Fax Server---> DID where > >> you have dialled originally. > >> > >> Cheers, > >> > >> George > >> > >> > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > >> >> > wrote: > >> > >> Did anyone see the issue or what I may have > >> missed in the config? As mentioned earlier, I > >> thought it would after a fresh install. Any help > >> is appreciated. > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Thanks for the quick response. Below is the > >> capture from the log viewer. In this test, I am > >> using a remote fax machine (line from Cox with > >> physical machine) faxing into the "email server" > >> where I created an extension for fax-to-email. > >> Calling number is 1479-246-0547. > >> > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:99 > >> sofia/external/+14792460547 at flowroute.com > >> Standard DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> mod_sofia.c:396 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:559 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State Change > >> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:556 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate Change HANGUP > >> -> DOWN > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1510 Close Channel > >> sofia/external/+14792460547 at flowroute.com > >> [CS_DESTROY] > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1506 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Ended > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1488 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Locked, Waiting on > >> external entities > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:440 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change CS_REPORTING > >> -> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State > >> REPORTING going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:92 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> REPORTING, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:446 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_HANGUP -> CS_REPORTING > >> > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:599 Hangup Command > >> with no Session system(/usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 > >> name=+14792460547-2013-01-10-13-39-26 > >> messages='result: Disconnected after permitted > >> retries sender:ATW-4792460547 pages:0' > >> domain=192.168.4.15 caller_id_name='' > >> caller_id_number=): > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:48 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> HANGUP, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:552 Sending BYE to > >> sofia/external/+14792460547 at flowroute.com > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:503 Channel > >> sofia/external/+14792460547 at flowroute.com > >> hanging up, > >> cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:2687 > >> sofia/external/+14792460547 at flowroute.com > >> skip receive > >> message [APPLICATION_EXEC_COMPLETE] (channel is > >> hungup already) > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:3017 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [KILL] > >> 2013-01-10 13:39:52.216151 [NOTICE] > >> mod_dptools.c:1150 Hangup > >> sofia/external/+14792460547 at flowroute.com > >> [CS_EXECUTE] > >> [NORMAL_CLEARING] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:2994 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change ACTIVE -> HANGUP > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> hangup() > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_codec.c:244 > >> sofia/external/+14792460547 at flowroute.com > >> Restore > >> previous codec PCMU:0. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:516 remote model: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:515 remote vendor: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:514 remote country: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:513 ECM status off > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:509 Total fax pages: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:507 Pages transferred: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > >> Fax Ident > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:505 Remote station id: > >> ATW-4792460547 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:500 Fax processing not > >> successful - result (48) Disconnected after > >> permitted retries. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> 2013-01-10 13:39:38.766190 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:29.576127 [ERR] > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > >> must be between 20 and 10000 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxRateManagement:transferredTCF > >> m=image 31632 udptl t38 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completing][200] > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > >> sofia/external/+14792460547 at flowroute.com > >> Update Callee > >> ID to "+14792460547" <+14792460547> > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [calling][0] > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:2647 Local SDP: > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1379 Raw write codec activation > >> Success L16 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_core_codec.c:219 > >> sofia/external/+14792460547 at flowroute.com > >> Push codec L16:70 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1363 Raw read codec activation > >> Success L16 20000 > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > >> > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_ivr_play_say.c:1682 done playing file > >> silence_stream://2000 > >> 2013-01-10 13:39:26.256197 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> switch_ivr_play_say.c:1309 Codec Activated > >> L16 at 8000hz 1 channels 20ms > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> playback(silence_stream://2000) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38_request]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38_request=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38]=[true] > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completed][200] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.006127 [NOTICE] > >> mod_dptools.c:1176 Channel > >> [sofia/external/+14792460547 at flowroute.com > >> ] has been > >> answered > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_channel.c:3395 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change RINGING -> ACTIVE > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=sendrecv > >> a=ptime:20 > >> a=silenceSupp:off - - - - > >> a=fmtp:101 0-16 > >> a=rtpmap:101 telephone-event/8000 > >> a=rtpmap:0 PCMU/8000 > >> m=audio 27588 RTP/AVP 0 101 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_sofia.c:856 Local SDP > >> sofia/external/+14792460547 at flowroute.com > >> : > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:1046 STUN Success > >> [72.201.112.131]:[27588] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> answer() > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [api_hangup_hook]=[system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=${last_fax} messages='result: > >> ${fax_result_text} > >> sender:${fax_remote_station_id} > >> pages:${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='${caller_id_name}' > >> caller_id_number=${caller_id_number}] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\${last_fax} > >> messages='result: \${fax_result_text} > >> sender:\${fax_remote_station_id} > >> pages:\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\${caller_id_name}' > >> caller_id_number=\${caller_id_number}) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [call_direction]=[local] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(call_direction=local) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action hangup() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> playback(silence_stream://2000) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38_request=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action answer() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\\\${last_fax} > >> messages='result: \\\${fax_result_text} > >> sender:\\\${fax_remote_station_id} > >> pages:\\\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\\\${caller_id_name}' > >> caller_id_number=\\\${caller_id_number}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test Fax] destination_number(3150) =~ > >> /^3150$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Test Fax] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain_user] destination_number(3150) =~ > >> /^\*97$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain_user] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain] destination_number(3150) =~ > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [send_to_voicemail] > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->send_to_voicemail] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [extension-intercom] > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->extension-intercom] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [intercept-ext] destination_number(3150) > >> =~ /^\*\*(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->intercept-ext] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_return] destination_number(3150) =~ > >> /^\*69$|^lcr$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_return] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_privacy] destination_number(3150) =~ > >> /^\*67(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_privacy] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [global] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > >> break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->global] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [redial] destination_number(3150) =~ > >> /^(redial|\*870)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->redial] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [group-intercept] destination_number(3150) > >> =~ /^\*8$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->group-intercept] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> ANTI-Action > >> set(call_direction=local) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call-direction] ${call_direction}() =~ > >> /^(inbound|outbound|local)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call-direction] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [Flowroute.11d] destination_number(3150) > >> =~ /^\+?(\d{11})$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Flowroute.11d] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->3150 in context default > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:25.816280 [NOTICE] > >> switch_ivr.c:1781 Transfer > >> sofia/external/+14792460547 at flowroute.com > >> to > >> XML[3150 at default] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_ivr.c:1774 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_EXECUTE -> CS_ROUTING > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> transfer(3150 > >> XML default) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_channel.c:1135 EXPORT (export_vars) > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [outside_call]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(outside_call=true) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> transfer(3150 XML default) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] > >> destination_number(15203039810) =~ > >> /^15203039810$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] context(public) =~ /public/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->Test_Number] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_debug] ${call_debug}(false) =~ > >> /^true$/ break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->call_debug] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(outside_call=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [outside_call] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->outside_call] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->unloop] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->15203039810 in context public > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> References: <50FC354E.80206@gmail.com> Message-ID: <50FCE279.2070707@gmail.com> On 20.01.2013 22:00, Jo?o Mesquita wrote: > Mimiko, FS does not remove echo nor it should. You need to start > providing more information on the problems you report on the mailing > list otherwise the first response will always be a request for more > information. > > Most probably, if you are hearing echo from your own voice, it must be > produced on your endpoint. My best guess would be acoustic echo produced > by the handset/headset or if you are using hands-free without decent AEC... > > What endpoint are you using? Softphone, hard phone or what? Dear Jo?o, As I write it's not phones fault of echo. If you are interesting, phone is DPH-150S from D-Link. This phone allow two SIP accounts. One is registered to FS and another to my phone provider. Same phone, two accounts. When using phone provider's account there is no echo. When using FS account, there is echo. So the handset does not have any problem. And I am not talking about hands-free. I did provide info for start. And I am asking where to dig to find the problem. If FS does not remove echo, then who does? In VoIP-FS direct connection who removes echo then? Thank you. -- Mimiko desu. From jalsot at gmail.com Mon Jan 21 10:25:03 2013 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Mon, 21 Jan 2013 08:25:03 +0100 Subject: [Freeswitch-users] Remove inband DTMF on record_session In-Reply-To: References: Message-ID: Hello Moy, Inserting silence instead of the tone would be fine for me. I think the >= 50ms delay in recording is not an issue. I guess, removing from "live" streams would be harder and would introduce the delay you mentioned. So you have a plan to add such a functionality in the near future? That would be amazing :) Regards, Jalsot On Mon, Jan 21, 2013 at 2:48 AM, Moises Silva wrote: > On Sun, Jan 20, 2013 at 4:12 PM, Tamas Jalsovszky wrote: > >> Hello, >> >> Is there a way to to remove inbad DTMF from the recorded wav file >> (conditionaly)? >> If not, how could be that done the easyiest way and a way it could get >> into FS git? >> >> As FS has a great DTMF detector and it has probably all the stuff doing >> such a feature, I think it would be much better than using an external tool >> for removing DTMF from finished wav file - and it would consume more I/O >> probably. >> >> > Removing it would not be that hard (ie, inserting silence instead). > Removing it without leaving a trace any DTMF was ever there sounds a tad > more difficult. If you're ok with inserting a silence instead of the tone, > extending the record_session app to detect and remove DTMF is probably what > I'd do. > > It is also possible to create a new app that monitors the audio stream and > silence the stream when DTMF is detected, but since the detection takes a > few milliseconds (~50ms), there would be a bit of "bleeding" of DTMF in the > signal, unless the new app also delays the signal by 50ms and then has a > way to strip the DTMF completely. > > *Moises Silva > **Manager, Software Engineering*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > ** > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter`| > | YouTube > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/68f70952/attachment.html From dvl36.ripe.nick at gmail.com Mon Jan 21 10:56:17 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Mon, 21 Jan 2013 09:56:17 +0200 Subject: [Freeswitch-users] Echo present. In-Reply-To: <50FCE279.2070707@gmail.com> References: <50FC354E.80206@gmail.com> <50FCE279.2070707@gmail.com> Message-ID: The problem with other side phone or PSTN gateway. Different delays in voice path, so in one case AEC algo work well, but not for another. WBR. 2013/1/21 Mimiko > On 20.01.2013 22:00, Jo?o Mesquita wrote: > > Mimiko, FS does not remove echo nor it should. You need to start > > providing more information on the problems you report on the mailing > > list otherwise the first response will always be a request for more > > information. > > > > Most probably, if you are hearing echo from your own voice, it must be > > produced on your endpoint. My best guess would be acoustic echo produced > > by the handset/headset or if you are using hands-free without decent > AEC... > > > > What endpoint are you using? Softphone, hard phone or what? > > Dear Jo?o, > > As I write it's not phones fault of echo. If you are interesting, phone > is DPH-150S from D-Link. This phone allow two SIP accounts. One is > registered to FS and another to my phone provider. Same phone, two > accounts. When using phone provider's account there is no echo. When > using FS account, there is echo. So the handset does not have any > problem. And I am not talking about hands-free. > > I did provide info for start. And I am asking where to dig to find the > problem. If FS does not remove echo, then who does? In VoIP-FS direct > connection who removes echo then? > > Thank you. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/de6f37e6/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Jan 21 14:34:20 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 21 Jan 2013 11:34:20 +0000 Subject: [Freeswitch-users] mod_xml_curl - request for comments In-Reply-To: References: <804D48104511D4468F0D60DF9D3100350B0A1102@MAIL.millicorp.com> Message-ID: Agreed, I'll get this done sometime this week. Thanks Cal On Fri, Jan 18, 2013 at 11:10 PM, Michael Collins wrote: > Anything that is globally applicable probably shouldn't be documented > solely on this page. It would be better to codify that in a separate page > that can be linked-to from various other pages. I would cut & paste that > information into a diff wiki page and link to it from the xml curl page and > see what that looks like. > > -MC > > > On Fri, Jan 18, 2013 at 9:34 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> I agree that having some alternative input/ideas on this page would >> certainly be a good thing. >> >> I'm almost tempted to say that this page doesn't need to explain/detail >> any of the channel variable contexts (the ones where it explains Applicable >> common fields etc), as this is really FreeSWITCH wide, >> and interpretation of these variables mostly depends on your own business >> logic. >> >> Removing this info would make the page a lot more read-able, but at the >> same time, might make it more difficult to understand for new users. >> >> Perhaps a warning at the top somewhere explaining this might be more >> suitable? >> >> Cal >> >> On Fri, Jan 18, 2013 at 4:49 PM, Michael Collins wrote: >> >>> Agreed, this is a significant improvement. Keep plugging away and if >>> anyone else has the time and knowledge to add to this page that would be >>> fantastic. >>> -MC >>> >>> On Fri, Jan 18, 2013 at 6:50 AM, Tim Meade wrote: >>> >>>> Hi Cal?.**** >>>> >>>> ** ** >>>> >>>> I think it looks MUCH better. **** >>>> >>>> ** ** >>>> >>>> Great job,**** >>>> >>>> ** ** >>>> >>>> Tim**** >>>> >>>> ** ** >>>> >>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal >>>> Leeming [Simplicity Media Ltd] >>>> *Sent:* Friday, January 18, 2013 9:33 AM >>>> *To:* FreeSWITCH Users Help >>>> *Subject:* [Freeswitch-users] mod_xml_curl - request for comments**** >>>> >>>> ** ** >>>> >>>> Hello,**** >>>> >>>> ** ** >>>> >>>> I've been re-reading my changes to mod_xml_curl documentation, and I'm >>>> still not entirely happy with it.**** >>>> >>>> ** ** >>>> >>>> I can't tell if it genuinely needs a different approach, or if I'm just >>>> being fussy.**** >>>> >>>> ** ** >>>> >>>> Therefore, constructive criticism is very much welcome - even if it >>>> means the whole damn thing has to be re-done again.**** >>>> >>>> ** ** >>>> >>>> http://wiki.freeswitch.org/wiki/Mod_xml_curl**** >>>> >>>> ** ** >>>> >>>> Thanks**** >>>> >>>> >>>> Cal**** >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/5286bfe3/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Mon Jan 21 14:36:42 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 21 Jan 2013 11:36:42 +0000 Subject: [Freeswitch-users] mod_xml_curl - request for comments In-Reply-To: References: <804D48104511D4468F0D60DF9D3100350B0A1102@MAIL.millicorp.com> Message-ID: Ah, I see what you've done now - that's actually a good idea. I'll have a tinker around and see what I can come up with - thank you for filling some of the TODOs! Cal On Mon, Jan 21, 2013 at 12:41 AM, Jo?o Mesquita wrote: > Cal, I filled in a couple of TODO you had there. Take a look. On the > directory, it might be best to separate requests as initiated by module so > that they are more clear. What do you think? Take a look at what I did and > see if it is clear to you. > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Fri, Jan 18, 2013 at 8:10 PM, Michael Collins wrote: > >> Anything that is globally applicable probably shouldn't be documented >> solely on this page. It would be better to codify that in a separate page >> that can be linked-to from various other pages. I would cut & paste that >> information into a diff wiki page and link to it from the xml curl page and >> see what that looks like. >> >> -MC >> >> >> On Fri, Jan 18, 2013 at 9:34 AM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> I agree that having some alternative input/ideas on this page would >>> certainly be a good thing. >>> >>> I'm almost tempted to say that this page doesn't need to explain/detail >>> any of the channel variable contexts (the ones where it explains Applicable >>> common fields etc), as this is really FreeSWITCH wide, >>> and interpretation of these variables mostly depends on your own business >>> logic. >>> >>> Removing this info would make the page a lot more read-able, but at the >>> same time, might make it more difficult to understand for new users. >>> >>> Perhaps a warning at the top somewhere explaining this might be more >>> suitable? >>> >>> Cal >>> >>> On Fri, Jan 18, 2013 at 4:49 PM, Michael Collins wrote: >>> >>>> Agreed, this is a significant improvement. Keep plugging away and if >>>> anyone else has the time and knowledge to add to this page that would be >>>> fantastic. >>>> -MC >>>> >>>> On Fri, Jan 18, 2013 at 6:50 AM, Tim Meade wrote: >>>> >>>>> Hi Cal?.**** >>>>> >>>>> ** ** >>>>> >>>>> I think it looks MUCH better. **** >>>>> >>>>> ** ** >>>>> >>>>> Great job,**** >>>>> >>>>> ** ** >>>>> >>>>> Tim**** >>>>> >>>>> ** ** >>>>> >>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal >>>>> Leeming [Simplicity Media Ltd] >>>>> *Sent:* Friday, January 18, 2013 9:33 AM >>>>> *To:* FreeSWITCH Users Help >>>>> *Subject:* [Freeswitch-users] mod_xml_curl - request for comments**** >>>>> >>>>> ** ** >>>>> >>>>> Hello,**** >>>>> >>>>> ** ** >>>>> >>>>> I've been re-reading my changes to mod_xml_curl documentation, and I'm >>>>> still not entirely happy with it.**** >>>>> >>>>> ** ** >>>>> >>>>> I can't tell if it genuinely needs a different approach, or if I'm >>>>> just being fussy.**** >>>>> >>>>> ** ** >>>>> >>>>> Therefore, constructive criticism is very much welcome - even if it >>>>> means the whole damn thing has to be re-done again.**** >>>>> >>>>> ** ** >>>>> >>>>> http://wiki.freeswitch.org/wiki/Mod_xml_curl**** >>>>> >>>>> ** ** >>>>> >>>>> Thanks**** >>>>> >>>>> >>>>> Cal**** >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/3b053b62/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Jan 21 14:42:48 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 21 Jan 2013 11:42:48 +0000 Subject: [Freeswitch-users] mod_xml_cdr and transfers - missing call legs In-Reply-To: References: Message-ID: Yeah, this was down to a silly mistake on my part during debugging, which I managed to repeat several times without realizing. Enabling B legs works exactly as expected, the hold_events are showing perfectly, I am getting a record for each new channel created by transfers, and linking them together is fairly easy. Thanks for the follow up! Cal On Mon, Jan 21, 2013 at 12:12 AM, Jo?o Mesquita wrote: > Cal, I saw from Jira that you were receiving the CDRs for all legs, > correct? B legs should always link to their A legs and that's how you find > what happened and when. Were you able to? > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Fri, Jan 18, 2013 at 8:18 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello, >> >> So, we are currently testing mod_xml_cdr and having some difficulties >> getting all the correct data. >> >> I actually have a fairly well commented dump of this happening (albeit >> from an unrelated bug); >> >> http://jira.freeswitch.org/secure/attachment/17781/debug-18-01-2013.txt >> >> Take the following scenario: >> >> From user/2000 TO user/2001 >> from user/2001 to user/2002 via user/2000 >> from user/2002 to gw/447534971120 via user/2001 >> from gw/447534971120 to ivr/2003 via user/2001 >> from gw/447534971120 to gw/442477062086 via ivr/2003 >> HANGUP from gw/447534971120 >> >> At this point the CDRs are posted back to our web app (both A and B >> leg).. however, it doesn't seem to contain all the above call events.. >> >> The end goal for me was to have each stage of a call logged and linked.. >> so if a call is transferred 10 times, I will see 10 CDR legs. >> >> So, my questions are; >> >> * Is this behaviour expected? >> * Am I approaching this from the wrong angle? >> * Are there better ways to collect CDRs? >> >> Any and all feedback would be much appreciated. >> >> Thanks >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/a4ea1d31/attachment-0001.html From matthias.cramer at iway.ch Mon Jan 21 18:19:39 2013 From: matthias.cramer at iway.ch (Matthias Cramer) Date: Mon, 21 Jan 2013 16:19:39 +0100 Subject: [Freeswitch-users] Make a call with Perl Message-ID: <50FD5C8B.9030306@iway.ch> Hi All I try to make a Perl Script which calls a certain external Phone number. It then should wait until someone picks up and play a sound file The user on the other end then needs to confirm the call with a pin Then another sound file should be played and the the call should be terminated. I tried several ways with ESL and originate. But I allays end up with a call I have no control over. I can read DTMF but can not hang up the call or play a second file. I think I am going in to a wrong direction. Any hints ? Regards Matthias -- Matthias Cramer / mc322-ripe Senior Network & Security Engineer iway AG Phone +41 43 500 1111 Badenerstrasse 569 Fax +41 44 271 3535 CH-8048 Z?rich http://www.iway.ch/ GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 253 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/8c0ff52f/attachment.bin From ben at langfeld.co.uk Mon Jan 21 20:27:37 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Mon, 21 Jan 2013 17:27:37 +0000 Subject: [Freeswitch-users] Make a call with Perl In-Reply-To: <50FD5C8B.9030306@iway.ch> References: <50FD5C8B.9030306@iway.ch> Message-ID: You could try out Adhearsion (http://adhearsion.com). Everything you mention is easily done there. Regards, Ben Langfeld On 21 January 2013 15:19, Matthias Cramer wrote: > Hi All > > I try to make a Perl Script which calls a certain external Phone number. > It then should wait until someone picks up and play a sound file > The user on the other end then needs to confirm the call with a pin > Then another sound file should be played and the the call should be > terminated. > > I tried several ways with ESL and originate. But I allays end up with a > call I have no > control over. I can read DTMF but can not hang up the call or play a > second file. > > I think I am going in to a wrong direction. > > Any hints ? > > Regards > > Matthias > > -- > Matthias Cramer / mc322-ripe Senior Network & Security Engineer > iway AG Phone +41 43 500 1111 > Badenerstrasse 569 Fax +41 44 271 3535 > CH-8048 Z?rich http://www.iway.ch/ > GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/fea62620/attachment.html From vbvbrj at gmail.com Mon Jan 21 20:43:43 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 21 Jan 2013 19:43:43 +0200 Subject: [Freeswitch-users] Reload mod_lua Message-ID: <50FD7E4F.6080607@gmail.com> I want to reload mod_lua in order to reload lua scripts that a starte automatically on start-up. But I get this: freeswitch at internal> reload mod_lua +OK Reloading XML -ERR unloading module [Module is not unloadable] -ERR loading module [Module already loaded] [CRIT] switch_loadable_module.c:1451 Module is not unloadable. [WARNING] switch_loadable_module.c:1400 Module mod_lua Already Loaded! [INFO] mod_enum.c:872 ENUM Reloaded [INFO] switch_time.c:1165 Timezone reloaded 530 definitions How to refresh modified lua scripts without reloading FS with restart? -- Mimiko desu. From jmesquita at freeswitch.org Mon Jan 21 20:52:24 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 21 Jan 2013 14:52:24 -0300 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: <50F975E5.2040206@coppice.org> <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com> Message-ID: For it to create the folders, a patch would be needed. Not a hard one tho... Is this desirable? Jo?o Mesquita FreeSWITCH? Solutions On Mon, Jan 21, 2013 at 2:59 AM, Robert Craig wrote: > As far as I know, FS is running as root. Below are the permissions. > > <13>:ls -ld storage/fax/3150/inbox > > drw-rw-rw- 2 root root 4096 Jan 18 10:28 storage/fax/3150/inbox > > I've gone from the 'freeswitch' folder all the way into the inbox folder > verifying the same permissions down the line. Should it include 'execute' > permissions as well? Whats even more interesting is that when I create a > new fax extension, it doesn't create the folder, such as 3150. I have to > manually create it. > > > > Robert > > > ------------------------------ > Date: Sun, 20 Jan 2013 17:07:50 -0300 > From: jmesquita at freeswitch.org > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Robert, you clearly still have a permissions problem on this specific > folder (according to your dialplan): > /usr/local/freeswitch/storage/fax/3150/inbox/ > > What is the user FreeSWITCH is running under and what are the current > permissions on the folder? > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Sun, Jan 20, 2013 at 3:36 PM, Robert Craig wrote: > > George, > > I'm having a hard time understanding your dialplan. I'm not a FS guru > so my apologies. It looks like your dialplan is meant for an ATA. I am just > trying to get the email portion configured first; specifically inbound. > Once I resolve the inbound issues, I'll work on outbound and then move to > ATA's. > > > Robert > > > ------------------------------ > Date: Sun, 20 Jan 2013 09:52:11 -0500 > From: gassaad at emassembly.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Here is mine: > Inbound: > http://pastebin.freeswitch.org/20471 > Outbound: > http://pastebin.freeswitch.org/20470 > > Thanks, > George > > On Sat, Jan 19, 2013 at 2:54 PM, Robert Craig wrote: > > OK, I copied everything from the fax extension dial plan into the below > pastebin. > > http://pastebin.freeswitch.org/20466 > > > Robert > > > ------------------------------ > From: bdfoster at endigotech.com > Date: Fri, 18 Jan 2013 12:59:38 -0500 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Submit the dialplan you are using to receive the fax. You can use the > FreeSWITCH pastebin, http://pastebin.freeswitch.org > > Sent from my iPhone > > On Jan 18, 2013, at 12:43 PM, Robert Craig > wrote: > > I believe I'm just using the built in SpanDSP for faxing. I've looked > around for 'rxfax' and can't find anything. I did notice that the directory > where the faxes were supposed to be stored for the extension I create > didn't even exist. I created them, applied the permissions, but it seems > like I am missing something because the same error is still there. > > > Robert > > > ------------------------------ > Date: Fri, 18 Jan 2013 09:13:36 -0800 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If you are using the rxfax app then you specify the file name and location: > > > > Make sure that the target directory has read/write permissions. You can > use something like "chmod 666 /my/directory" to make it read/write. > > -MC > > On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: > > Where does the file get downloaded to? What should the permissions be? > > Robert > > > > Date: Sat, 19 Jan 2013 00:18:45 +0800 > > From: steveu at coppice.org > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] T38 Configuration > > > > If the TIFF file cannot be opened on receive you probably have a file > > permissions issue (e.g. with the directory in which the file is to be > > created). If the TIFF file cannot be opened on transmit there are > > various possibilities about the suitability of the file's contents. > > > > Steve > > > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > > This was an incoming fax. I don't have much control over the other > > > end. Why is that error appearing when trying to receive something? > > > > > > Robert > > > > > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > > wrote: > > > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > > >> Please try pdf, you should be in business now. > > >> Please let us know, > > >> > > >> On Jan 18, 2013 12:35 AM, "Robert Craig" > >> > wrote: > > >> > > >> OK, so I've made a little bit of progress. The distant end is a > > >> Windows Fax server hooked up to an analog line. I created a fax > > >> extension in FS, assigned an inbound route to the extension, and > > >> made sure the extension had an email address. I believe the fax > > >> receive wants to complete, but there is an interesting error in > > >> the log about TIF files. Anyone ever seen this before? > > >> > > >> v=0 > > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> c=IN IP4 72.201.112.131 > > >> t=0 0 > > >> m=image 27714 udptl t38 > > >> a=T38FaxVersion:0 > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxFillBitRemoval > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > > >> ip/port confirmed. > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > > >> Negotiation Result > > >> ======================================================= > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > > >> status off > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > > >> country: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > > >> vendor: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > > >> model: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > > >> processing not successful - result (41) TIFF/F file cannot be > > >> opened.* > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > > >> transferred: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > > >> fax pages: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > > >> resolution: 0x0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > > >> status off > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > > >> country: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > > >> vendor: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > > >> model: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > > >> sofia/external/+14792460547 @flowroute.com > > >> Restore previous codec PCMU:0. > > >> EXECUTE sofia/external/+14792460547 > > >> @flowroute.com hangup() > > >> > > >> Robert > > >> > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> OK, I am not extremely concerned with getting an ATA working > > >> right now. If I could just get an inbound extension-to-email > > >> working it would be great. I'll start from scratch and then let > > >> you know what happens. Just to make sure, I am going to create an > > >> extension, assign an email address to it, make sure there is an > > >> inbound route to that extension, and then test it. > > >> > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > > >> From: gassaad at emassembly.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> The reason for outbound extension is: > > >> When you send a fax from ATA to the outside world, it will dial > > >> first to the outbound extension and then after successful receive > > >> it will dial out to the intended number. > > >> > > >> As for the error you have it may be related to your ATA. I am not > > >> sure please see why do you get the following message. > > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > > >> > > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > > >> > wrote: > > >> > > >> OK, I've already tried creating one extension for inbound and > > >> set an email address. Of course, it failed and the errors are > > >> at the bottom of this email. What is the purpose of the > > >> outbound extension? > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > > >> From: gassaad at emassembly.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Sorry for the delay to get back to you. > > >> 1- App --> fax server > > >> 2- Create two extensions (one for inbound) another for > > >> (outbound) From within either extension you can send faxes > > >> "virtual fax" > > >> 3- point your create your outbound and inbound dialplans as > > >> normal but make it to point to those extensions. > > >> > > >> Good luck, > > >> > > >> George > > >> > > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > > >> > > > >> wrote: > > >> > > >> What did you do on the config side to get it to work? > > >> I've tried both an Ata and a fax to email extension and > > >> they both fail in both directions. > > >> > > >> Robert > > >> > > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > > >> > > > >> wrote: > > >> > > >> I use Fusion successfully to send / receive fax. > > >> The way it works: > > >> ATA --->FaxServer. > > >> After successful receipt, Fax Server---> DID where > > >> you have dialled originally. > > >> > > >> Cheers, > > >> > > >> George > > >> > > >> > > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > > >> > >> > wrote: > > >> > > >> Did anyone see the issue or what I may have > > >> missed in the config? As mentioned earlier, I > > >> thought it would after a fresh install. Any help > > >> is appreciated. > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Thanks for the quick response. Below is the > > >> capture from the log viewer. In this test, I am > > >> using a remote fax machine (line from Cox with > > >> physical machine) faxing into the "email server" > > >> where I created an extension for fax-to-email. > > >> Calling number is 1479-246-0547. > > >> > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:99 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> mod_sofia.c:396 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:559 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State Change > > >> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:556 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate Change HANGUP > > >> -> DOWN > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1510 Close Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_DESTROY] > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1506 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Ended > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1488 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Locked, Waiting on > > >> external entities > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:440 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change CS_REPORTING > > >> -> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State > > >> REPORTING going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:92 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> REPORTING, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:446 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_HANGUP -> CS_REPORTING > > >> > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:599 Hangup Command > > >> with no Session system(/usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 > > >> name=+14792460547-2013-01-10-13-39-26 > > >> messages='result: Disconnected after permitted > > >> retries sender:ATW-4792460547 pages:0' > > >> domain=192.168.4.15 caller_id_name='' > > >> caller_id_number=): > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:48 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> HANGUP, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:552 Sending BYE to > > >> sofia/external/+14792460547 at flowroute.com > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:503 Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> hanging up, > > >> cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:2687 > > >> sofia/external/+14792460547 at flowroute.com > > >> skip receive > > >> message [APPLICATION_EXEC_COMPLETE] (channel is > > >> hungup already) > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:3017 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [KILL] > > >> 2013-01-10 13:39:52.216151 [NOTICE] > > >> mod_dptools.c:1150 Hangup > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_EXECUTE] > > >> [NORMAL_CLEARING] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:2994 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change ACTIVE -> HANGUP > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> hangup() > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_codec.c:244 > > >> sofia/external/+14792460547 at flowroute.com > > >> Restore > > >> previous codec PCMU:0. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:516 remote model: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:515 remote vendor: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:514 remote country: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:513 ECM status off > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:509 Total fax pages: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:507 Pages transferred: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > > >> Fax Ident > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:505 Remote station id: > > >> ATW-4792460547 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:500 Fax processing not > > >> successful - result (48) Disconnected after > > >> permitted retries. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:38.766190 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:29.576127 [ERR] > > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > > >> must be between 20 and 10000 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxRateManagement:transferredTCF > > >> m=image 31632 udptl t38 > > >> t=0 0 > > >> c=IN IP4 66.42.121.5 > > >> s=- > > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > > >> v=0 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > > >> Remote SDP: > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completing][200] > > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > > >> sofia/external/+14792460547 at flowroute.com > > >> Update Callee > > >> ID to "+14792460547" <+14792460547> > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [calling][0] > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:2647 Local SDP: > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1379 Raw write codec activation > > >> Success L16 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_core_codec.c:219 > > >> sofia/external/+14792460547 at flowroute.com > > >> Push codec L16:70 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1363 Raw read codec activation > > >> Success L16 20000 > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > > >> > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_ivr_play_say.c:1682 done playing file > > >> silence_stream://2000 > > >> 2013-01-10 13:39:26.256197 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> switch_ivr_play_say.c:1309 Codec Activated > > >> L16 at 8000hz 1 channels 20ms > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> playback(silence_stream://2000) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38_request]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38_request=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38]=[true] > > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completed][200] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.006127 [NOTICE] > > >> mod_dptools.c:1176 Channel > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] has been > > >> answered > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_channel.c:3395 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change RINGING -> ACTIVE > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=sendrecv > > >> a=ptime:20 > > >> a=silenceSupp:off - - - - > > >> a=fmtp:101 0-16 > > >> a=rtpmap:101 telephone-event/8000 > > >> a=rtpmap:0 PCMU/8000 > > >> m=audio 27588 RTP/AVP 0 101 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_sofia.c:856 Local SDP > > >> sofia/external/+14792460547 at flowroute.com > > >> : > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:1046 STUN Success > > >> [72.201.112.131]:[27588] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> answer() > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [api_hangup_hook]=[system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=${last_fax} messages='result: > > >> ${fax_result_text} > > >> sender:${fax_remote_station_id} > > >> pages:${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='${caller_id_name}' > > >> caller_id_number=${caller_id_number}] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\${last_fax} > > >> messages='result: \${fax_result_text} > > >> sender:\${fax_remote_station_id} > > >> pages:\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\${caller_id_name}' > > >> caller_id_number=\${caller_id_number}) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-spymap/+14792460547 > /d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [call_direction]=[local] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(call_direction=local) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action hangup() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> playback(silence_stream://2000) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38_request=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action answer() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\\\${last_fax} > > >> messages='result: \\\${fax_result_text} > > >> sender:\\\${fax_remote_station_id} > > >> pages:\\\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\\\${caller_id_name}' > > >> caller_id_number=\\\${caller_id_number}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test Fax] destination_number(3150) =~ > > >> /^3150$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Test Fax] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain_user] destination_number(3150) =~ > > >> /^\*97$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain_user] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain] destination_number(3150) =~ > > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [send_to_voicemail] > > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->send_to_voicemail] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [extension-intercom] > > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->extension-intercom] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [intercept-ext] destination_number(3150) > > >> =~ /^\*\*(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->intercept-ext] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_return] destination_number(3150) =~ > > >> /^\*69$|^lcr$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_return] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_privacy] destination_number(3150) =~ > > >> /^\*67(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_privacy] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [global] > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > > >> break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->global] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [redial] destination_number(3150) =~ > > >> /^(redial|\*870)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->redial] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [group-intercept] destination_number(3150) > > >> =~ /^\*8$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->group-intercept] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> ANTI-Action > > >> set(call_direction=local) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call-direction] ${call_direction}() =~ > > >> /^(inbound|outbound|local)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call-direction] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [Flowroute.11d] destination_number(3150) > > >> =~ /^\+?(\d{11})$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Flowroute.11d] continue=false > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->3150 in context default > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [NOTICE] > > >> switch_ivr.c:1781 Transfer > > >> sofia/external/+14792460547 at flowroute.com > > >> to > > >> XML[3150 at default] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_ivr.c:1774 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_EXECUTE -> CS_ROUTING > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> transfer(3150 > > >> XML default) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_channel.c:1135 EXPORT (export_vars) > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [outside_call]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(outside_call=true) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> transfer(3150 XML default) > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] > > >> destination_number(15203039810) =~ > > >> /^15203039810$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] context(public) =~ /public/ > > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->Test_Number] continue=false > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_debug] ${call_debug}(false) =~ > > > >> /^true$/ break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->call_debug] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(outside_call=true) > > > >> > > ... > > [Message clipped] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/74233f27/attachment-0001.html From schoch+freeswitch.org at xwin32.com Mon Jan 21 21:00:07 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 21 Jan 2013 10:00:07 -0800 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: <50F975E5.2040206@coppice.org> <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com> Message-ID: On Sun, Jan 20, 2013 at 9:59 PM, Robert Craig wrote: > As far as I know, FS is running as root. Below are the permissions. > > <13>:ls -ld storage/fax/3150/inbox > > drw-rw-rw- 2 root root 4096 Jan 18 10:28 storage/fax/3150/inbox > > I've gone from the 'freeswitch' folder all the way into the inbox folder > verifying the same permissions down the line. Should it include 'execute' > permissions as well? > I'm a FreeSWITCH novice, but I've been using Unix systems since the early 1980's. Yes, the 'execute' permission needs to be set. The execute permission on a directory allows files (and directories) in that directory to be accessed. A directory with 'read', but not 'execute' permission allows you to see a list of files in that directory, but you can't access them or create new ones. It is very unusual to see a directory with the permissions you have shown. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/86df10a2/attachment.html From yehavi.bourvine at gmail.com Mon Jan 21 21:03:42 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 21 Jan 2013 20:03:42 +0200 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? Message-ID: Hello, We are running version 1.0.6; we did not upgrade for a long time due to some problems (mainly stuck BLFs) with the newer versions which have been corrected a few months ago. A few weeks ago I pulled the latest GIT and did functional testings on our development system; all was OK. Yesterday I moved our production system to that version. After a few hours the problems started: stuck BLFs, hangups, LUA scripts sometimes do not run. I had to revert back to the old version. My question: Is there some simple test suite which I can test various scnarios on my development system? like, "call XX, while this call is active call YY which is monitored by XX and hangup", etc. It is hard to cover all cases manually, so I would like something automated. Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/2e2cb178/attachment.html From msc at freeswitch.org Mon Jan 21 21:29:43 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 10:29:43 -0800 Subject: [Freeswitch-users] Cannot save Voicemail message In-Reply-To: <6770CA93-7D95-404A-809A-732A68C60771@kavun.ch> References: <3263505F-2AC8-4BB5-ABEB-44B706AD80C5@kavun.ch> <6770CA93-7D95-404A-809A-732A68C60771@kavun.ch> Message-ID: Indeed, no information at all. See if you can reproduce this on latest git HEAD. If you can, please open a Jira so that the devs can track this issue. Also, if this was working on a previous version of FreeSWITCH it would be very helpful to find out which commit broke it. -MC On Sat, Jan 19, 2013 at 4:31 PM, Emrah wrote: > The console logs are not very helpful? > > http://pastebin.freeswitch.org/20467 > > Thanks for any help. > > E > On Jan 18, 2013, at 6:24 PM, Michael Collins wrote: > > > Can you supply a bit more context? Show us more log lines. > > -MC > > > > On Fri, Jan 18, 2013 at 2:18 PM, Emrah wrote: > > Hi all, > > > > I've noticed a strange problem lately, I can't press 3 to save a VM. > > > > I use mod_shout, my VMs are in MP3 format. > > > > Log abstract: > > > > 2013-01-18 17:11:09.191700 [DEBUG] switch_ivr_play_say.c:1309 Codec > Activated L16 at 16000hz 1 channels 20ms > > 2013-01-18 17:11:10.191704 [DEBUG] switch_rtp.c:3827 RTP RECV DTMF 3:1120 > > 2013-01-18 17:11:10.191704 [DEBUG] switch_ivr_play_say.c:1682 done > playing file > /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-save_recording.wav > > 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:67 No language > specified - Using [en] > > 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[voicemail/vm-you_have.wav] (en:en) > > > > freeswitch at internal> version > > FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 > 2012-12-29 00:17:59Z) > > > > Deleting works fine. > > Any idea? > > > > Thanks and all the best, > > Emrah > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/e8cfdb11/attachment.html From msc at freeswitch.org Mon Jan 21 21:36:19 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 10:36:19 -0800 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: That may be the case, but that doesn't actually supply the information you need to find out where the DTMFs are coming from. I would recommend doing the test that Moy suggested. At least find out where FreeSWITCH thinks this DTMF is coming from. -MC On Sun, Jan 20, 2013 at 6:43 PM, Subhash wrote: > But we don't see any problem when both endpoints are SIP. > > Thanks, > Subhash. > > > On Mon, Jan 21, 2013 at 7:12 AM, Moises Silva wrote: > >> On Fri, Jan 18, 2013 at 1:27 AM, Subhash wrote: >> >>> Yes,we are trying to bridge PSTN and VOIP call. >>> >>> We posted the log in pastebin.Please use the below URL to access the >>> freeswitch log. >>> >>> http://pastebin.freeswitch.org/20461 >>> >>> >> >> >> 1. 013-01-16 01:02:26.954165 [INFO] ftmod_sangoma_isdn_stack_rcv.c:144 >> [s4c1][4:1] Received CONNECT/CONNECT ACK (suId:1 suInstId:2 spInstId: >> 2 ces:0) >> 2. 2013-01-16 01:02:26.954165 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c: >> 285 [s4c1][4:1] Processing CONNECT/CONNECT ACK (suId:1 suInstId:2 >> spInstId:2 ces:0) >> 3. 2013-01-16 01:02:26.974165 [DEBUG] switch_rtp.c:3596 Correct >> ip/port confirmed. >> 4. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:1002 [s4c1][4:1] First >> packet read stats: Rx queue len: 4, Rx queue size: 10 >> 5. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:938 [s4c1][4:1] First >> packet write stats: Tx queue len: 1, Tx queue size: 5, Tx idle: 10 >> 6. 2013-01-16 01:02:27.694163 [DEBUG] switch_rtp.c:3797 RTP RECV DTMF >> 9:640 >> 7. 2013-01-16 01:02:27.694163 [DEBUG] switch_ivr_bridge.c:393 Send >> signal FreeTDM/4:1/2001 [BREAK] >> 8. 2013-01-16 01:02:27.694163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >> DTMF [9] >> >> >> The DTMF comes from the RTP side, not FreeTDM (as shown by the log >> above). You have to verify using tcpdump/wireshark if there is any real >> DTMF being received. >> >> I'd do exactly the same test again, but this time I would also start a >> wireshark trace. Then please pastebin the trace again and upload the pcap >> trace to cloudshark (http://www.cloudshark.org/) and paste the link here. >> >> Moy >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/f5b596e3/attachment-0001.html From vbvbrj at gmail.com Mon Jan 21 21:36:43 2013 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 21 Jan 2013 20:36:43 +0200 Subject: [Freeswitch-users] Echo present. In-Reply-To: References: <50FC354E.80206@gmail.com> <50FCE279.2070707@gmail.com> Message-ID: <50FD8ABB.4010809@gmail.com> On 21.01.2013 09:56, Dmitry Lysenko wrote: > The problem with other side phone or PSTN gateway. > Different delays in voice path, so in one case AEC algo work well, but > not for another. > WBR. I made calls to different types of networks: PSTN, VoIP, GSM. About 50% I hear echo. Including calling to same number consequently, echo ay be present or not. -- Mimiko desu. From slava at tangramltd.com Mon Jan 21 21:43:57 2013 From: slava at tangramltd.com (Viacheslav Dubrovskyi) Date: Mon, 21 Jan 2013 20:43:57 +0200 Subject: [Freeswitch-users] How to enable debug mode for DTMF in FreeTDM? Message-ID: <50FD8C6D.3090806@tangramltd.com> Hi. How to enable debug mode for DTMF in FreeTDM? I setup in freetdm.conf [general] debugdtmf_directory=/var/spool/freeswitch/dtmf [span zt mySpan] debugdtmf => yes But there is nothing in path. What else should I do? -- WBR, Viacheslav Dubrovskyi -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3746 bytes Desc: ?????????????????????????????????? ?????????????? S/MIME Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/af507b56/attachment.bin From msc at freeswitch.org Mon Jan 21 21:44:17 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 10:44:17 -0800 Subject: [Freeswitch-users] Echo present. In-Reply-To: <50FD8ABB.4010809@gmail.com> References: <50FC354E.80206@gmail.com> <50FCE279.2070707@gmail.com> <50FD8ABB.4010809@gmail.com> Message-ID: Echo should be removed at the endpoints, not in the middle of the stream. In most cases, if you hear echo then the other end's echo canceller is not working, and vice versa. One thing you can do is tcpdump or pcapsipdump the traffic coming in on the wire and listen to the audio in Wireshark's player tool. (If the codec is PCMU or PCMA.) If the echo you hear is in the RTP stream that comes in on the wire then you know the echo can on the far end is not working very well. -MC On Mon, Jan 21, 2013 at 10:36 AM, Mimiko wrote: > On 21.01.2013 09:56, Dmitry Lysenko wrote: > > The problem with other side phone or PSTN gateway. > > Different delays in voice path, so in one case AEC algo work well, but > > not for another. > > WBR. > > I made calls to different types of networks: PSTN, VoIP, GSM. About 50% > I hear echo. Including calling to same number consequently, echo ay be > present or not. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/808e5bdd/attachment.html From msc at freeswitch.org Mon Jan 21 21:52:59 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 10:52:59 -0800 Subject: [Freeswitch-users] Reload mod_lua In-Reply-To: <50FD7E4F.6080607@gmail.com> References: <50FD7E4F.6080607@gmail.com> Message-ID: Lua is not "unloadable" and therefore you cannot do this the way you are trying to. You'll need to restart FreeSWITCH. -MC On Mon, Jan 21, 2013 at 9:43 AM, Mimiko wrote: > I want to reload mod_lua in order to reload lua scripts that a starte > automatically on start-up. But I get this: > > freeswitch at internal> reload mod_lua > +OK Reloading XML > -ERR unloading module [Module is not unloadable] > -ERR loading module [Module already loaded] > > [CRIT] switch_loadable_module.c:1451 Module is not unloadable. > [WARNING] switch_loadable_module.c:1400 Module mod_lua Already Loaded! > [INFO] mod_enum.c:872 ENUM Reloaded > [INFO] switch_time.c:1165 Timezone reloaded 530 definitions > > How to refresh modified lua scripts without reloading FS with restart? > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/1642527e/attachment.html From msc at freeswitch.org Mon Jan 21 22:00:35 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 11:00:35 -0800 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: Message-ID: This test suite does not presently exist. I would love to see some work done in this area. One challenge is that there are numerous kinds of devices that can interop with FS so the test suite will need to have some way to handle the various phone makes and models. Also, I don't know which hard phones, if any, support automated testing. I would *love* to see some sort of web/network API on various phones where you could poll the phone to see the state of each SLA/BLF key as well as what's in the display. That would make automated testing much cleaner. As it stands now, testing BLFs is a tedious process of watching phones and waiting to see if the lights do what they should be doing. If anyone has practical suggestions (nothing pie-in-the-sky, if you get my meaning) then by all means share with the group. -MC On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine wrote: > Hello, > > We are running version 1.0.6; we did not upgrade for a long time due to > some problems (mainly stuck BLFs) with the newer versions which have been > corrected a few months ago. > > A few weeks ago I pulled the latest GIT and did functional testings on our > development system; all was OK. > > Yesterday I moved our production system to that version. After a few hours > the problems started: stuck BLFs, hangups, LUA scripts sometimes do not > run. I had to revert back to the old version. > > My question: Is there some simple test suite which I can test various > scnarios on my development system? like, "call XX, while this call is > active call YY which is monitored by XX and hangup", etc. It is hard to > cover all cases manually, so I would like something automated. > > Thanks, __Yehavi: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/fedb6fd4/attachment-0001.html From gabe at gundy.org Mon Jan 21 22:26:31 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 21 Jan 2013 12:26:31 -0700 Subject: [Freeswitch-users] Accessing SIP Replaces Header In-Reply-To: References: Message-ID: On Sat, Jan 19, 2013 at 10:50 PM, afshin afzali wrote: > I use Lua's session:getVariable to get it, also using uuid_dump to list all > available variables. That's what I was wondering. If you used uuid_dump (and you did it on the right channel), I don't have anything more to offer. Sorry. Best, Gabe From bdfoster at endigotech.com Mon Jan 21 22:27:37 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 21 Jan 2013 14:27:37 -0500 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: <50F975E5.2040206@coppice.org> <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com> Message-ID: Why are you running Freeswitch as root? Sent from my iPhone On Jan 21, 2013, at 12:59 AM, Robert Craig wrote: > As far as I know, FS is running as root. Below are the permissions. > > <13>:ls -ld storage/fax/3150/inbox > > drw-rw-rw- 2 root root 4096 Jan 18 10:28 storage/fax/3150/inbox > > I've gone from the 'freeswitch' folder all the way into the inbox folder verifying the same permissions down the line. Should it include 'execute' permissions as well? Whats even more interesting is that when I create a new fax extension, it doesn't create the folder, such as 3150. I have to manually create it. > > > > Robert > > > > Date: Sun, 20 Jan 2013 17:07:50 -0300 > From: jmesquita at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Robert, you clearly still have a permissions problem on this specific folder (according to your dialplan): > /usr/local/freeswitch/storage/fax/3150/inbox/ > > What is the user FreeSWITCH is running under and what are the current permissions on the folder? > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Sun, Jan 20, 2013 at 3:36 PM, Robert Craig wrote: > George, > > I'm having a hard time understanding your dialplan. I'm not a FS guru so my apologies. It looks like your dialplan is meant for an ATA. I am just trying to get the email portion configured first; specifically inbound. Once I resolve the inbound issues, I'll work on outbound and then move to ATA's. > > > Robert > > > Date: Sun, 20 Jan 2013 09:52:11 -0500 > From: gassaad at emassembly.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Here is mine: > Inbound: > http://pastebin.freeswitch.org/20471 > Outbound: > http://pastebin.freeswitch.org/20470 > > Thanks, > George > > On Sat, Jan 19, 2013 at 2:54 PM, Robert Craig wrote: > OK, I copied everything from the fax extension dial plan into the below pastebin. > > http://pastebin.freeswitch.org/20466 > > > Robert > > > From: bdfoster at endigotech.com > Date: Fri, 18 Jan 2013 12:59:38 -0500 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Submit the dialplan you are using to receive the fax. You can use the FreeSWITCH pastebin, http://pastebin.freeswitch.org > > Sent from my iPhone > > On Jan 18, 2013, at 12:43 PM, Robert Craig wrote: > > I believe I'm just using the built in SpanDSP for faxing. I've looked around for 'rxfax' and can't find anything. I did notice that the directory where the faxes were supposed to be stored for the extension I create didn't even exist. I created them, applied the permissions, but it seems like I am missing something because the same error is still there. > > > Robert > > > Date: Fri, 18 Jan 2013 09:13:36 -0800 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If you are using the rxfax app then you specify the file name and location: > > > > Make sure that the target directory has read/write permissions. You can use something like "chmod 666 /my/directory" to make it read/write. > > -MC > > On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: > Where does the file get downloaded to? What should the permissions be? > > Robert > > > > Date: Sat, 19 Jan 2013 00:18:45 +0800 > > From: steveu at coppice.org > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] T38 Configuration > > > > If the TIFF file cannot be opened on receive you probably have a file > > permissions issue (e.g. with the directory in which the file is to be > > created). If the TIFF file cannot be opened on transmit there are > > various possibilities about the suitability of the file's contents. > > > > Steve > > > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > > This was an incoming fax. I don't have much control over the other > > > end. Why is that error appearing when trying to receive something? > > > > > > Robert > > > > > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > > wrote: > > > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > > >> Please try pdf, you should be in business now. > > >> Please let us know, > > >> > > >> On Jan 18, 2013 12:35 AM, "Robert Craig" > >> > wrote: > > >> > > >> OK, so I've made a little bit of progress. The distant end is a > > >> Windows Fax server hooked up to an analog line. I created a fax > > >> extension in FS, assigned an inbound route to the extension, and > > >> made sure the extension had an email address. I believe the fax > > >> receive wants to complete, but there is an interesting error in > > >> the log about TIF files. Anyone ever seen this before? > > >> > > >> v=0 > > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> c=IN IP4 72.201.112.131 > > >> t=0 0 > > >> m=image 27714 udptl t38 > > >> a=T38FaxVersion:0 > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxFillBitRemoval > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > > >> ip/port confirmed. > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > > >> Negotiation Result > > >> ======================================================= > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > > >> status off > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > > >> country: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > > >> vendor: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > > >> model: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > > >> ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > > >> ============================================================================== > > >> > > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > > >> processing not successful - result (41) TIFF/F file cannot be > > >> opened.* > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > > >> transferred: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > > >> fax pages: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > > >> resolution: 0x0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > > >> status off > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > > >> country: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > > >> vendor: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > > >> model: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > > >> ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > > >> sofia/external/+14792460547 @flowroute.com > > >> Restore previous codec PCMU:0. > > >> EXECUTE sofia/external/+14792460547 > > >> @flowroute.com hangup() > > >> > > >> Robert > > >> > > >> > > >> > > >> ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> OK, I am not extremely concerned with getting an ATA working > > >> right now. If I could just get an inbound extension-to-email > > >> working it would be great. I'll start from scratch and then let > > >> you know what happens. Just to make sure, I am going to create an > > >> extension, assign an email address to it, make sure there is an > > >> inbound route to that extension, and then test it. > > >> > > >> > > >> Robert > > >> > > >> > > >> ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > > >> From: gassaad at emassembly.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> The reason for outbound extension is: > > >> When you send a fax from ATA to the outside world, it will dial > > >> first to the outbound extension and then after successful receive > > >> it will dial out to the intended number. > > >> > > >> As for the error you have it may be related to your ATA. I am not > > >> sure please see why do you get the following message. > > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > > >> > > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > > >> > wrote: > > >> > > >> OK, I've already tried creating one extension for inbound and > > >> set an email address. Of course, it failed and the errors are > > >> at the bottom of this email. What is the purpose of the > > >> outbound extension? > > >> > > >> Robert > > >> > > >> > > >> ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > > >> From: gassaad at emassembly.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Sorry for the delay to get back to you. > > >> 1- App --> fax server > > >> 2- Create two extensions (one for inbound) another for > > >> (outbound) From within either extension you can send faxes > > >> "virtual fax" > > >> 3- point your create your outbound and inbound dialplans as > > >> normal but make it to point to those extensions. > > >> > > >> Good luck, > > >> > > >> George > > >> > > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > > >> > > > >> wrote: > > >> > > >> What did you do on the config side to get it to work? > > >> I've tried both an Ata and a fax to email extension and > > >> they both fail in both directions. > > >> > > >> Robert > > >> > > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > > >> > > > >> wrote: > > >> > > >> I use Fusion successfully to send / receive fax. > > >> The way it works: > > >> ATA --->FaxServer. > > >> After successful receipt, Fax Server---> DID where > > >> you have dialled originally. > > >> > > >> Cheers, > > >> > > >> George > > >> > > >> > > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > > >> > >> > wrote: > > >> > > >> Did anyone see the issue or what I may have > > >> missed in the config? As mentioned earlier, I > > >> thought it would after a fresh install. Any help > > >> is appreciated. > > >> > > >> Robert > > >> > > >> > > >> ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Thanks for the quick response. Below is the > > >> capture from the log viewer. In this test, I am > > >> using a remote fax machine (line from Cox with > > >> physical machine) faxing into the "email server" > > >> where I created an extension for fax-to-email. > > >> Calling number is 1479-246-0547. > > >> > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:99 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> mod_sofia.c:396 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:559 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State Change > > >> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:556 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate Change HANGUP > > >> -> DOWN > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1510 Close Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_DESTROY] > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1506 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Ended > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1488 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Locked, Waiting on > > >> external entities > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:440 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change CS_REPORTING > > >> -> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State > > >> REPORTING going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:92 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> REPORTING, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:446 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_HANGUP -> CS_REPORTING > > >> > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:599 Hangup Command > > >> with no Session system(/usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 > > >> name=+14792460547-2013-01-10-13-39-26 > > >> messages='result: Disconnected after permitted > > >> retries sender:ATW-4792460547 pages:0' > > >> domain=192.168.4.15 caller_id_name='' > > >> caller_id_number=): > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:48 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> HANGUP, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:552 Sending BYE to > > >> sofia/external/+14792460547 at flowroute.com > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:503 Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> hanging up, > > >> cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:2687 > > >> sofia/external/+14792460547 at flowroute.com > > >> skip receive > > >> message [APPLICATION_EXEC_COMPLETE] (channel is > > >> hungup already) > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:3017 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [KILL] > > >> 2013-01-10 13:39:52.216151 [NOTICE] > > >> mod_dptools.c:1150 Hangup > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_EXECUTE] > > >> [NORMAL_CLEARING] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:2994 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change ACTIVE -> HANGUP > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> hangup() > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_codec.c:244 > > >> sofia/external/+14792460547 at flowroute.com > > >> Restore > > >> previous codec PCMU:0. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:518 > > >> ============================================================================== > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:516 remote model: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:515 remote vendor: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:514 remote country: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:513 ECM status off > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:509 Total fax pages: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:507 Pages transferred: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > > >> Fax Ident > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:505 Remote station id: > > >> ATW-4792460547 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:500 Fax processing not > > >> successful - result (48) Disconnected after > > >> permitted retries. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:487 > > >> ============================================================================== > > >> > > >> 2013-01-10 13:39:38.766190 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:29.576127 [ERR] > > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > > >> must be between 20 and 10000 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxRateManagement:transferredTCF > > >> m=image 31632 udptl t38 > > >> t=0 0 > > >> c=IN IP4 66.42.121.5 > > >> s=- > > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > > >> v=0 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > > >> Remote SDP: > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completing][200] > > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > > >> sofia/external/+14792460547 at flowroute.com > > >> Update Callee > > >> ID to "+14792460547" <+14792460547> > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [calling][0] > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:2647 Local SDP: > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1379 Raw write codec activation > > >> Success L16 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_core_codec.c:219 > > >> sofia/external/+14792460547 at flowroute.com > > >> Push codec L16:70 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1363 Raw read codec activation > > >> Success L16 20000 > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > > >> > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_ivr_play_say.c:1682 done playing file > > >> silence_stream://2000 > > >> 2013-01-10 13:39:26.256197 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> switch_ivr_play_say.c:1309 Codec Activated > > >> L16 at 8000hz 1 channels 20ms > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> playback(silence_stream://2000) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38_request]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38_request=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38]=[true] > > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completed][200] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.006127 [NOTICE] > > >> mod_dptools.c:1176 Channel > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] has been > > >> answered > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_channel.c:3395 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change RINGING -> ACTIVE > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=sendrecv > > >> a=ptime:20 > > >> a=silenceSupp:off - - - - > > >> a=fmtp:101 0-16 > > >> a=rtpmap:101 telephone-event/8000 > > >> a=rtpmap:0 PCMU/8000 > > >> m=audio 27588 RTP/AVP 0 101 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_sofia.c:856 Local SDP > > >> sofia/external/+14792460547 at flowroute.com > > >> : > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:1046 STUN Success > > >> [72.201.112.131]:[27588] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> answer() > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [api_hangup_hook]=[system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=${last_fax} messages='result: > > >> ${fax_result_text} > > >> sender:${fax_remote_station_id} > > >> pages:${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='${caller_id_name}' > > >> caller_id_number=${caller_id_number}] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\${last_fax} > > >> messages='result: \${fax_result_text} > > >> sender:\${fax_remote_station_id} > > >> pages:\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\${caller_id_name}' > > >> caller_id_number=\${caller_id_number}) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [call_direction]=[local] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(call_direction=local) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action hangup() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> playback(silence_stream://2000) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38_request=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action answer() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\\\${last_fax} > > >> messages='result: \\\${fax_result_text} > > >> sender:\\\${fax_remote_station_id} > > >> pages:\\\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\\\${caller_id_name}' > > >> caller_id_number=\\\${caller_id_number}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test Fax] destination_number(3150) =~ > > >> /^3150$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Test Fax] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain_user] destination_number(3150) =~ > > >> /^\*97$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain_user] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain] destination_number(3150) =~ > > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [send_to_voicemail] > > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->send_to_voicemail] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [extension-intercom] > > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->extension-intercom] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [intercept-ext] destination_number(3150) > > >> =~ /^\*\*(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->intercept-ext] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_return] destination_number(3150) =~ > > >> /^\*69$|^lcr$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_return] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_privacy] destination_number(3150) =~ > > >> /^\*67(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_privacy] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [global] > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > > >> break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->global] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [redial] destination_number(3150) =~ > > >> /^(redial|\*870)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->redial] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [group-intercept] destination_number(3150) > > >> =~ /^\*8$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->group-intercept] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> ANTI-Action > > >> set(call_direction=local) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call-direction] ${call_direction}() =~ > > >> /^(inbound|outbound|local)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call-direction] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [Flowroute.11d] destination_number(3150) > > >> =~ /^\+?(\d{11})$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Flowroute.11d] continue=false > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->3150 in context default > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [NOTICE] > > >> switch_ivr.c:1781 Transfer > > >> sofia/external/+14792460547 at flowroute.com > > >> to > > >> XML[3150 at default] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_ivr.c:1774 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_EXECUTE -> CS_ROUTING > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> transfer(3150 > > >> XML default) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_channel.c:1135 EXPORT (export_vars) > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [outside_call]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(outside_call=true) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> transfer(3150 XML default) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] > > >> destination_number(15203039810) =~ > > >> /^15203039810$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] context(public) =~ /public/ > > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->Test_Number] continue=false > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_debug] ${call_debug}(false) =~ > > > >> /^true$/ break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->call_debug] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(outside_call=true) > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [outside_call] > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->outside_call] continue=true > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > > >> Regex > > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->unloop] continue=false > > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->15203039810 in context public > > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> ... > > [Message clipped] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/a78de808/attachment-0001.html From msc at freeswitch.org Mon Jan 21 22:42:42 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 11:42:42 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Hello to our great OSS community! We are happy to report that the first ClueCon 2012 videos have been posted to our YouTube channel ! The first talks posted are: * I'm Living In The Wrong Future!- Dr. Moshe Yudkowsky * ClueCon 2012 Keynote Address - Anthony Minessale This week we will be posting a few more videos from day one of ClueCon 2012. Join this week's conference call or subscribe to our YouTube channel to find out which videos are uploaded next. Now that calendar year 2013 is in full swing we are making plans for ClueCon 2013, which will be held on August 6-8, 2013. Once again we'll be at the Hyatt Chicago Magnificent Mile . Stay tuned for more information about this year's event. The development team has been working hard on some cool new things which will be announced in due course. In the meantime Ken Rice is working very hard on getting FreeSWITCH 1.2.6 ready for final release. If you have a sandbox system that you can use to test the latest git HEAD please do so. Also, feel free to visit our bug tracker and see if there are any bugs that you can fix or verify as was discussed on last week's conference call. This week we will be having an open discussionon topics that are dear to you. Please bring your questions for discussion and any tips and tricks that you'd like to share with the community. Have a great week! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/8c06feb3/attachment.html From msc at freeswitch.org Mon Jan 21 22:50:11 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 11:50:11 -0800 Subject: [Freeswitch-users] Make a call with Perl In-Reply-To: <50FD5C8B.9030306@iway.ch> References: <50FD5C8B.9030306@iway.ch> Message-ID: If you have the FS Cookbook then check out chapter 4, page 81. If you don't have the FS Cookbook then *SHAME* on you! :D Tip: you can download the examples from both books at the Packt Publishing website. The only caveat as far as I can tell is that you need to create a login account. Grab the examples for the Cookbook and look at chapter 4. The recipe is called "Launching a call with an inbound event socket connect" and it's written in - you guessed it - Perl. :) -MC On Mon, Jan 21, 2013 at 7:19 AM, Matthias Cramer wrote: > Hi All > > I try to make a Perl Script which calls a certain external Phone number. > It then should wait until someone picks up and play a sound file > The user on the other end then needs to confirm the call with a pin > Then another sound file should be played and the the call should be > terminated. > > I tried several ways with ESL and originate. But I allays end up with a > call I have no > control over. I can read DTMF but can not hang up the call or play a > second file. > > I think I am going in to a wrong direction. > > Any hints ? > > Regards > > Matthias > > -- > Matthias Cramer / mc322-ripe Senior Network & Security Engineer > iway AG Phone +41 43 500 1111 > Badenerstrasse 569 Fax +41 44 271 3535 > CH-8048 Z?rich http://www.iway.ch/ > GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/0c4d5d07/attachment.html From mario_fs at mgtech.com Mon Jan 21 22:54:30 2013 From: mario_fs at mgtech.com (Mario G) Date: Mon, 21 Jan 2013 11:54:30 -0800 Subject: [Freeswitch-users] adding In-Reply-To in sip header not working In-Reply-To: References: Message-ID: Thanks, apparently I had it wrong, the doc below states that the PBX must support it incoming, they pointed me to using effective_caller_id which I added to the bridge but it still does not work. Would love to fix this since the cell phones currently have no idea who is calling. Mario G Please note that this feature is ONLY AVAILABLE for customers using a SIP PBX that either supports (or allows the configuring of) the "in-reply-to" header (defined by RFC 3261) for incoming calls which are forwarded to an outbound trunk. In these instances Callcentric will "Pass-Through" the CallerID from the original call which was received to the outbound bridged/forwarded call. On Jan 19, 2013, at 5:13 PM, Richard Brady wrote: > On 20 January 2013 00:06, Mario G wrote: > I never did this so I must be missing something, I tried both below but the bridge then fails. Can anyone shed some light on what I am doing wrong. My ITSP now supports in-reply-to so I can pass the caller ID to a forwarded call from FS. > > In-Reply-To should contain a Call-ID not a caller ID. They are very different. > > The following would make a bit more sense, but still not a lot: > > > > Using In-Reply-To in a response doesn't seem right to me. I would expect it to appear in an INVITE. For for example, you get a missed call and you call the person back, then the INVITE for the callback would have a new Call-ID but the original Call-ID in the In-Reply-To header. That said, I have no idea what your ITSPs intended use for the header is. > > Richard > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/69977ff0/attachment.html From freeswitch at orresta.no-ip.com Mon Jan 21 23:55:22 2013 From: freeswitch at orresta.no-ip.com (Jakob) Date: Mon, 21 Jan 2013 21:55:22 +0100 Subject: [Freeswitch-users] Portaudio troubles Message-ID: <50FDAB3A.5050602@orresta.no-ip.com> Hi, I'm trying to get working a line into en external PA system Whenever i run this extension Freeswitch coredumps and has to be restarted. The log for the last call in the console before it goes dead follows EXECUTE sofia/external/1050 at rshaninge.mress.se bridge(portaudio/endpoint/out-right) 2013-01-21 21:46:45.057266 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:2362 Created shared audio stream alpha: 48000 channels 2 2013-01-21 21:46:45.576716 [NOTICE] switch_channel.c:968 New Channel portaudio/endpoint-out-right [d9f50f6b-ea97-47a9-b721-659c1a40af14] 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:1272 (portaudio/endpoint-out-right) State Change CS_NEW -> CS_INIT 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:1283 Send signal portaudio/endpoint-out-right [BREAK] 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:804 portaudio/endpoint-out-right CHANNEL KILL 2013-01-21 21:46:45.576716 [DEBUG] switch_core_state_machine.c:415 (portaudio/endpoint-out-right) Running State Change CS_INIT 2013-01-21 21:46:45.576716 [DEBUG] switch_core_state_machine.c:454 (portaudio/endpoint-out-right) State INIT 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:286 (portaudio/endpoint-out-right) State Change CS_INIT -> CS_ROUTING 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:1283 Send signal portaudio/endpoint-out-right [BREAK] 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:804 portaudio/endpoint-out-right CHANNEL KILL 2013-01-21 21:46:45.576716 [DEBUG] switch_core_state_machine.c:454 (portaudio/endpoint-out-right) State INIT going to sleep 2013-01-21 21:46:45.576716 [DEBUG] switch_core_state_machine.c:415 (portaudio/endpoint-out-right) Running State Change CS_ROUTING 2013-01-21 21:46:45.576716 [DEBUG] switch_channel.c:2003 (portaudio/endpoint-out-right) Callstate Change DOWN -> RINGING 2013-01-21 21:46:45.576716 [DEBUG] switch_core_state_machine.c:470 (portaudio/endpoint-out-right) State ROUTING 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:666 Added call 1 2013-01-21 21:46:45.576716 [DEBUG] switch_channel.c:3395 (portaudio/endpoint-out-right) Callstate Change RINGING -> ACTIVE 2013-01-21 21:46:45.576716 [DEBUG] switch_channel.c:3407 Send signal sofia/external/1050 at rshaninge.mress.se [BREAK] 2013-01-21 21:46:45.576716 [NOTICE] mod_portaudio.c:360 Channel [portaudio/endpoint-out-right] has been answered 2013-01-21 21:46:45.576716 [DEBUG] sofia_glue.c:3351 AUDIO RTP [sofia/external/1050 at rshaninge.mress.se] 81.216.195.38 port 24486 -> 10.171.149.115 port 21000 codec: 9 ms: 20 2013-01-21 21:46:45.576716 [DEBUG] switch_rtp.c:1940 Starting timer [soft] 160 bytes per 20ms 2013-01-21 21:46:45.576716 [DEBUG] sofia_glue.c:3615 Set 2833 dtmf send payload to 101 2013-01-21 21:46:45.576716 [DEBUG] sofia_glue.c:3621 Set 2833 dtmf receive payload to 101 2013-01-21 21:46:45.576716 [DEBUG] sofia_glue.c:3648 sofia/external/1050 at rshaninge.mress.se Set rtp dtmf delay to 40 2013-01-21 21:46:45.576716 [DEBUG] mod_sofia.c:856 Local SDP sofia/external/1050 at rshaninge.mress.se: v=0 o=FreeSWITCH 1358776719 1358776720 IN IP4 81.216.195.38 s=FreeSWITCH c=IN IP4 81.216.195.38 t=0 0 m=audio 24486 RTP/AVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:830 Send signal sofia/external/1050 at rshaninge.mress.se [BREAK] 2013-01-21 21:46:45.576716 [DEBUG] switch_channel.c:3395 (sofia/external/1050 at rshaninge.mress.se) Callstate Change RINGING -> ACTIVE 2013-01-21 21:46:45.576716 [NOTICE] switch_ivr_originate.c:3378 Channel [sofia/external/1050 at rshaninge.mress.se] has been answered 2013-01-21 21:46:45.576716 [DEBUG] switch_ivr_originate.c:3435 Originate Resulted in Success: [portaudio/endpoint-out-right] 2013-01-21 21:46:45.576716 [DEBUG] switch_ivr_bridge.c:1268 (portaudio/endpoint-out-right) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:1283 Send signal portaudio/endpoint-out-right [BREAK] 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:804 portaudio/endpoint-out-right CHANNEL KILL 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:830 Send signal portaudio/endpoint-out-right [BREAK] 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:804 portaudio/endpoint-out-right CHANNEL KILL 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:830 Send signal sofia/external/1050 at rshaninge.mress.se [BREAK] 2013-01-21 21:46:45.576716 [DEBUG] switch_ivr_bridge.c:1370 (portaudio/endpoint-out-right) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:1283 Send signal portaudio/endpoint-out-right [BREAK] 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:804 portaudio/endpoint-out-right CHANNEL KILL 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:975 Send signal sofia/external/1050 at rshaninge.mress.se [BREAK] 2013-01-21 21:46:45.576716 [DEBUG] sofia.c:5599 Channel sofia/external/1050 at rshaninge.mress.se entering state [completed][200] 2013-01-21 21:46:45.636815 [DEBUG] switch_core_session.c:975 Send signal sofia/external/1050 at rshaninge.mress.se [BREAK] 2013-01-21 21:46:45.636815 [DEBUG] switch_core_session.c:975 Send signal sofia/external/1050 at rshaninge.mress.se [BREAK] 2013-01-21 21:46:45.636815 [DEBUG] switch_core_session.c:975 Send signal sofia/external/1050 at rshaninge.mress.se [BREAK] 2013-01-21 21:46:45.636815 [DEBUG] sofia.c:5599 Channel sofia/external/1050 at rshaninge.mress.se entering state [ready][200] 2013-01-21 21:46:46.576712 [DEBUG] mod_portaudio.c:429 portaudio/endpoint-out-right CHANNEL ROUTING 2013-01-21 21:46:46.576712 [DEBUG] switch_core_state_machine.c:470 (portaudio/endpoint-out-right) State ROUTING going to sleep 2013-01-21 21:46:46.576712 [DEBUG] switch_core_state_machine.c:415 (portaudio/endpoint-out-right) Running State Change CS_EXCHANGE_MEDIA 2013-01-21 21:46:46.576712 [DEBUG] switch_core_state_machine.c:480 (portaudio/endpoint-out-right) State EXCHANGE_MEDIA 2013-01-21 21:46:46.576712 [DEBUG] mod_portaudio.c:817 CHANNEL LOOPBACK 2013-01-21 21:46:46.576712 [DEBUG] switch_core_session.c:892 Send signal portaudio/endpoint-out-right [BREAK] 2013-01-21 21:46:46.576712 [DEBUG] mod_portaudio.c:804 portaudio/endpoint-out-right CHANNEL KILL 2013-01-21 21:46:46.576712 [DEBUG] switch_core_session.c:892 Send signal sofia/external/1050 at rshaninge.mress.se [BREAK] Am I missing something in the log that helps in tracking this down. Best Regards Jakob Sundberg From craig.robert at hotmail.com Tue Jan 22 00:10:52 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Mon, 21 Jan 2013 14:10:52 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: , , , , , , , , , , , , , <50F975E5.2040206@coppice.org>, , , , <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com>, , , , , Message-ID: OK, so I'll change the permissions to include read/write/execute. FS is running as root because this is a box at the house I am playing with. I installed FS using one of the script methods out there and I was logged in as root at the time when I ran everything. Regardless, I'm not a fan of the script install because as you can see, a simple feature such as faxing has created headaches. I didn't think it would be this difficult to turn on fax to email after a default install out of the box. My Asterisk box has given me nothing but headaches when it comes to fax signals so I figured I would give FS a try in that aspect. Once I change the permissions, I'll reboot the thing just to make sure and then run another test. If this works, I am going to completely rebuild this thing and hopefully the folder creation problem goes away as well. When I get to that point, what source/guide do you guys recommend for a manual install? Robert From: bdfoster at endigotech.com Date: Mon, 21 Jan 2013 14:27:37 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Why are you running Freeswitch as root? Sent from my iPhone On Jan 21, 2013, at 12:59 AM, Robert Craig wrote: As far as I know, FS is running as root. Below are the permissions. <13>:ls -ld storage/fax/3150/inbox drw-rw-rw- 2 root root 4096 Jan 18 10:28 storage/fax/3150/inbox I've gone from the 'freeswitch' folder all the way into the inbox folder verifying the same permissions down the line. Should it include 'execute' permissions as well? Whats even more interesting is that when I create a new fax extension, it doesn't create the folder, such as 3150. I have to manually create it. Robert Date: Sun, 20 Jan 2013 17:07:50 -0300 From: jmesquita at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Robert, you clearly still have a permissions problem on this specific folder (according to your dialplan):/usr/local/freeswitch/storage/fax/3150/inbox/ What is the user FreeSWITCH is running under and what are the current permissions on the folder? Jo?o Mesquita FreeSWITCH? Solutions On Sun, Jan 20, 2013 at 3:36 PM, Robert Craig wrote: George, I'm having a hard time understanding your dialplan. I'm not a FS guru so my apologies. It looks like your dialplan is meant for an ATA. I am just trying to get the email portion configured first; specifically inbound. Once I resolve the inbound issues, I'll work on outbound and then move to ATA's. Robert Date: Sun, 20 Jan 2013 09:52:11 -0500 From: gassaad at emassembly.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Here is mine: Inbound:http://pastebin.freeswitch.org/20471Outbound:http://pastebin.freeswitch.org/20470 Thanks,George On Sat, Jan 19, 2013 at 2:54 PM, Robert Craig wrote: OK, I copied everything from the fax extension dial plan into the below pastebin. http://pastebin.freeswitch.org/20466 Robert From: bdfoster at endigotech.com Date: Fri, 18 Jan 2013 12:59:38 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Submit the dialplan you are using to receive the fax. You can use the FreeSWITCH pastebin, http://pastebin.freeswitch.org Sent from my iPhone On Jan 18, 2013, at 12:43 PM, Robert Craig wrote: I believe I'm just using the built in SpanDSP for faxing. I've looked around for 'rxfax' and can't find anything. I did notice that the directory where the faxes were supposed to be stored for the extension I create didn't even exist. I created them, applied the permissions, but it seems like I am missing something because the same error is still there. Robert Date: Fri, 18 Jan 2013 09:13:36 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration If you are using the rxfax app then you specify the file name and location: Make sure that the target directory has read/write permissions. You can use something like "chmod 666 /my/directory" to make it read/write. -MC On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: Where does the file get downloaded to? What should the permissions be? Robert > Date: Sat, 19 Jan 2013 00:18:45 +0800 > From: steveu at coppice.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If the TIFF file cannot be opened on receive you probably have a file > permissions issue (e.g. with the directory in which the file is to be > created). If the TIFF file cannot be opened on transmit there are > various possibilities about the suitability of the file's contents. > > Steve > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > This was an incoming fax. I don't have much control over the other > > end. Why is that error appearing when trying to receive something? > > > > Robert > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > wrote: > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > >> Please try pdf, you should be in business now. > >> Please let us know, > >> > >> On Jan 18, 2013 12:35 AM, "Robert Craig" >> > wrote: > >> > >> OK, so I've made a little bit of progress. The distant end is a > >> Windows Fax server hooked up to an analog line. I created a fax > >> extension in FS, assigned an inbound route to the extension, and > >> made sure the extension had an email address. I believe the fax > >> receive wants to complete, but there is an interesting error in > >> the log about TIF files. Anyone ever seen this before? > >> > >> v=0 > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> c=IN IP4 72.201.112.131 > >> t=0 0 > >> m=image 27714 udptl t38 > >> a=T38FaxVersion:0 > >> a=T38MaxBitRate:9600 > >> a=T38FaxFillBitRemoval > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxUdpEC:t38UDPRedundancy > >> > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > >> ip/port confirmed. > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > >> Negotiation Result > >> ======================================================= > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > >> status off > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > >> country: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > >> vendor: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > >> model: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > >> processing not successful - result (41) TIFF/F file cannot be > >> opened.* > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > >> transferred: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > >> fax pages: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > >> resolution: 0x0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > >> status off > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > >> country: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > >> vendor: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > >> model: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > >> sofia/external/+14792460547 @flowroute.com > >> Restore previous codec PCMU:0. > >> EXECUTE sofia/external/+14792460547 > >> @flowroute.com hangup() > >> > >> Robert > >> > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> OK, I am not extremely concerned with getting an ATA working > >> right now. If I could just get an inbound extension-to-email > >> working it would be great. I'll start from scratch and then let > >> you know what happens. Just to make sure, I am going to create an > >> extension, assign an email address to it, make sure there is an > >> inbound route to that extension, and then test it. > >> > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > >> From: gassaad at emassembly.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> The reason for outbound extension is: > >> When you send a fax from ATA to the outside world, it will dial > >> first to the outbound extension and then after successful receive > >> it will dial out to the intended number. > >> > >> As for the error you have it may be related to your ATA. I am not > >> sure please see why do you get the following message. > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > >> > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > >> > wrote: > >> > >> OK, I've already tried creating one extension for inbound and > >> set an email address. Of course, it failed and the errors are > >> at the bottom of this email. What is the purpose of the > >> outbound extension? > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > >> From: gassaad at emassembly.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Sorry for the delay to get back to you. > >> 1- App --> fax server > >> 2- Create two extensions (one for inbound) another for > >> (outbound) From within either extension you can send faxes > >> "virtual fax" > >> 3- point your create your outbound and inbound dialplans as > >> normal but make it to point to those extensions. > >> > >> Good luck, > >> > >> George > >> > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > >> > > >> wrote: > >> > >> What did you do on the config side to get it to work? > >> I've tried both an Ata and a fax to email extension and > >> they both fail in both directions. > >> > >> Robert > >> > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > >> > > >> wrote: > >> > >> I use Fusion successfully to send / receive fax. > >> The way it works: > >> ATA --->FaxServer. > >> After successful receipt, Fax Server---> DID where > >> you have dialled originally. > >> > >> Cheers, > >> > >> George > >> > >> > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > >> >> > wrote: > >> > >> Did anyone see the issue or what I may have > >> missed in the config? As mentioned earlier, I > >> thought it would after a fresh install. Any help > >> is appreciated. > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Thanks for the quick response. Below is the > >> capture from the log viewer. In this test, I am > >> using a remote fax machine (line from Cox with > >> physical machine) faxing into the "email server" > >> where I created an extension for fax-to-email. > >> Calling number is 1479-246-0547. > >> > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:99 > >> sofia/external/+14792460547 at flowroute.com > >> Standard DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> mod_sofia.c:396 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:559 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State Change > >> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:556 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate Change HANGUP > >> -> DOWN > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1510 Close Channel > >> sofia/external/+14792460547 at flowroute.com > >> [CS_DESTROY] > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1506 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Ended > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1488 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Locked, Waiting on > >> external entities > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:440 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change CS_REPORTING > >> -> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State > >> REPORTING going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:92 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> REPORTING, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:446 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_HANGUP -> CS_REPORTING > >> > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:599 Hangup Command > >> with no Session system(/usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 > >> name=+14792460547-2013-01-10-13-39-26 > >> messages='result: Disconnected after permitted > >> retries sender:ATW-4792460547 pages:0' > >> domain=192.168.4.15 caller_id_name='' > >> caller_id_number=): > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:48 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> HANGUP, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:552 Sending BYE to > >> sofia/external/+14792460547 at flowroute.com > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:503 Channel > >> sofia/external/+14792460547 at flowroute.com > >> hanging up, > >> cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:2687 > >> sofia/external/+14792460547 at flowroute.com > >> skip receive > >> message [APPLICATION_EXEC_COMPLETE] (channel is > >> hungup already) > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:3017 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [KILL] > >> 2013-01-10 13:39:52.216151 [NOTICE] > >> mod_dptools.c:1150 Hangup > >> sofia/external/+14792460547 at flowroute.com > >> [CS_EXECUTE] > >> [NORMAL_CLEARING] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:2994 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change ACTIVE -> HANGUP > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> hangup() > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_codec.c:244 > >> sofia/external/+14792460547 at flowroute.com > >> Restore > >> previous codec PCMU:0. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:516 remote model: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:515 remote vendor: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:514 remote country: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:513 ECM status off > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:509 Total fax pages: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:507 Pages transferred: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > >> Fax Ident > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:505 Remote station id: > >> ATW-4792460547 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:500 Fax processing not > >> successful - result (48) Disconnected after > >> permitted retries. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> 2013-01-10 13:39:38.766190 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:29.576127 [ERR] > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > >> must be between 20 and 10000 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxRateManagement:transferredTCF > >> m=image 31632 udptl t38 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completing][200] > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > >> sofia/external/+14792460547 at flowroute.com > >> Update Callee > >> ID to "+14792460547" <+14792460547> > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [calling][0] > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:2647 Local SDP: > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1379 Raw write codec activation > >> Success L16 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_core_codec.c:219 > >> sofia/external/+14792460547 at flowroute.com > >> Push codec L16:70 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1363 Raw read codec activation > >> Success L16 20000 > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > >> > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_ivr_play_say.c:1682 done playing file > >> silence_stream://2000 > >> 2013-01-10 13:39:26.256197 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> switch_ivr_play_say.c:1309 Codec Activated > >> L16 at 8000hz 1 channels 20ms > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> playback(silence_stream://2000) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38_request]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38_request=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38]=[true] > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completed][200] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.006127 [NOTICE] > >> mod_dptools.c:1176 Channel > >> [sofia/external/+14792460547 at flowroute.com > >> ] has been > >> answered > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_channel.c:3395 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change RINGING -> ACTIVE > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=sendrecv > >> a=ptime:20 > >> a=silenceSupp:off - - - - > >> a=fmtp:101 0-16 > >> a=rtpmap:101 telephone-event/8000 > >> a=rtpmap:0 PCMU/8000 > >> m=audio 27588 RTP/AVP 0 101 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_sofia.c:856 Local SDP > >> sofia/external/+14792460547 at flowroute.com > >> : > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:1046 STUN Success > >> [72.201.112.131]:[27588] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> answer() > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [api_hangup_hook]=[system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=${last_fax} messages='result: > >> ${fax_result_text} > >> sender:${fax_remote_station_id} > >> pages:${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='${caller_id_name}' > >> caller_id_number=${caller_id_number}] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\${last_fax} > >> messages='result: \${fax_result_text} > >> sender:\${fax_remote_station_id} > >> pages:\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\${caller_id_name}' > >> caller_id_number=\${caller_id_number}) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [call_direction]=[local] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(call_direction=local) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action hangup() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> playback(silence_stream://2000) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38_request=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action answer() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\\\${last_fax} > >> messages='result: \\\${fax_result_text} > >> sender:\\\${fax_remote_station_id} > >> pages:\\\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\\\${caller_id_name}' > >> caller_id_number=\\\${caller_id_number}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test Fax] destination_number(3150) =~ > >> /^3150$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Test Fax] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain_user] destination_number(3150) =~ > >> /^\*97$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain_user] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain] destination_number(3150) =~ > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [send_to_voicemail] > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->send_to_voicemail] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [extension-intercom] > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->extension-intercom] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [intercept-ext] destination_number(3150) > >> =~ /^\*\*(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->intercept-ext] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_return] destination_number(3150) =~ > >> /^\*69$|^lcr$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_return] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_privacy] destination_number(3150) =~ > >> /^\*67(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_privacy] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [global] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > >> break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->global] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [redial] destination_number(3150) =~ > >> /^(redial|\*870)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->redial] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [group-intercept] destination_number(3150) > >> =~ /^\*8$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->group-intercept] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> ANTI-Action > >> set(call_direction=local) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call-direction] ${call_direction}() =~ > >> /^(inbound|outbound|local)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call-direction] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [Flowroute.11d] destination_number(3150) > >> =~ /^\+?(\d{11})$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Flowroute.11d] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->3150 in context default > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:25.816280 [NOTICE] > >> switch_ivr.c:1781 Transfer > >> sofia/external/+14792460547 at flowroute.com > >> to > >> XML[3150 at default] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_ivr.c:1774 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_EXECUTE -> CS_ROUTING > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> transfer(3150 > >> XML default) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_channel.c:1135 EXPORT (export_vars) > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [outside_call]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(outside_call=true) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> transfer(3150 XML default) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] > >> destination_number(15203039810) =~ > >> /^15203039810$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] context(public) =~ /public/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->Test_Number] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_debug] ${call_debug}(false) =~ > >> /^true$/ break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->call_debug] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(outside_call=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [outside_call] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->outside_call] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->unloop] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->15203039810 in context public > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> References: <50FD7E4F.6080607@gmail.com> Message-ID: <50FDAFAE.8080608@gmail.com> or you can have your script listen for event or check a variable and exit the script as needed. and another background script which will start the script again if it's not running. not a simple solution but doable. On 1/21/2013 1:52 PM, Michael Collins wrote: > Lua is not "unloadable" and therefore you cannot do this the way you > are trying to. You'll need to restart FreeSWITCH. > -MC > > On Mon, Jan 21, 2013 at 9:43 AM, Mimiko > wrote: > > I want to reload mod_lua in order to reload lua scripts that a starte > automatically on start-up. But I get this: > > freeswitch at internal> reload mod_lua > +OK Reloading XML > -ERR unloading module [Module is not unloadable] > -ERR loading module [Module already loaded] > > [CRIT] switch_loadable_module.c:1451 Module is not unloadable. > [WARNING] switch_loadable_module.c:1400 Module mod_lua Already Loaded! > [INFO] mod_enum.c:872 ENUM Reloaded > [INFO] switch_time.c:1165 Timezone reloaded 530 definitions > > How to refresh modified lua scripts without reloading FS with restart? > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/5715d214/attachment.html From avi at avimarcus.net Tue Jan 22 00:21:16 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 21 Jan 2013 23:21:16 +0200 Subject: [Freeswitch-users] Reload mod_lua In-Reply-To: <50FDAFAE.8080608@gmail.com> References: <50FD7E4F.6080607@gmail.com> <50FDAFAE.8080608@gmail.com> Message-ID: A tool that can monitor Lua threads, and kill them, would come in useful every so often... -Avi On Mon, Jan 21, 2013 at 11:14 PM, Abaci wrote: > or you can have your script listen for event or check a variable and > exit the script as needed. and another background script which will start > the script again if it's not running. not a simple solution but doable. > > > On 1/21/2013 1:52 PM, Michael Collins wrote: > > Lua is not "unloadable" and therefore you cannot do this the way you are > trying to. You'll need to restart FreeSWITCH. > -MC > > On Mon, Jan 21, 2013 at 9:43 AM, Mimiko wrote: > >> I want to reload mod_lua in order to reload lua scripts that a starte >> automatically on start-up. But I get this: >> >> freeswitch at internal> reload mod_lua >> +OK Reloading XML >> -ERR unloading module [Module is not unloadable] >> -ERR loading module [Module already loaded] >> >> [CRIT] switch_loadable_module.c:1451 Module is not unloadable. >> [WARNING] switch_loadable_module.c:1400 Module mod_lua Already Loaded! >> [INFO] mod_enum.c:872 ENUM Reloaded >> [INFO] switch_time.c:1165 Timezone reloaded 530 definitions >> >> How to refresh modified lua scripts without reloading FS with restart? >> -- >> Mimiko desu. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/2d62d3d9/attachment.html From freeswitch at orresta.no-ip.com Tue Jan 22 00:22:08 2013 From: freeswitch at orresta.no-ip.com (Jakob) Date: Mon, 21 Jan 2013 22:22:08 +0100 Subject: [Freeswitch-users] anybody working on a swedish mod_say? In-Reply-To: References: <50D8D157.7020409@orresta.no-ip.com> Message-ID: <50FDB180.3010404@orresta.no-ip.com> Ok, I now have a full set of recordings for mod_say_sv and mod_voicemail. In what format should I submit them? 48k.wav? 32? 16? ulaw? and to where? I also have a working mod_say_sv module written with the english example as a base. So it is aware of the different encodings and not hardcoded to .wav files. This is now in production on my box and have been solid. There are however a patch to the FS-core that were needed. Adding a new gender to make the natural flow of recordings working. And a few patches to build files. I'm still working on getting the full voicemail system all swedish. How can I submit this for the benefit of others? Regards Jakob Sundberg 12/26/12 19:41, Michael Collins skrev: > Also, if you get a set of Swedish recordings let us know and we'll get > them hosted on our files.freeswitch.org > server. > -MC > > On Mon, Dec 24, 2012 at 2:17 PM, Avi Marcus > wrote: > > I can't find a list on the wiki.. but there doesn't seem to be a > mod_say_** for swedish prompts. > see in git source: src/mod/say > > If one of the languages is similar in making things > singular/plural, masculine/femanine you can use that as a template. > > Or you can create speech phrases for numbers with the logic in > there, but that's a hack.. > > > -Avi > > > On Tue, Dec 25, 2012 at 12:04 AM, Jakob > > wrote: > > Hi, > > I started working on a project that would include a voicemail > app. but I > have trouble finding a swedish implementation of a mod_say so > that I can > populate it with swedish prompts. > > I already have an working IVR, in swedish but there are still > a few > bugs, mainly getting english numbers in my swedish prompts. > > Jakob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/63148b1a/attachment-0001.html From craig.robert at hotmail.com Tue Jan 22 00:39:56 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Mon, 21 Jan 2013 14:39:56 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: , , , , , , , , , , , , , , , , , , , , , , , , , , <50F975E5.2040206@coppice.org>, , , , , , , , <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com>, , , , , , , , , , , Message-ID: It worked! The read/write/execute permissions fixed it. Now, oddly enough, the PDF didn't get emailed out. I have an email address assigned to the fax extension, but it apparently never sent out. I don't see anything in the logs. Can someone point me in the right direction on making it manually send, or configured the email server? Thanks. Robert From: craig.robert at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Mon, 21 Jan 2013 14:10:52 -0700 Subject: Re: [Freeswitch-users] T38 Configuration OK, so I'll change the permissions to include read/write/execute. FS is running as root because this is a box at the house I am playing with. I installed FS using one of the script methods out there and I was logged in as root at the time when I ran everything. Regardless, I'm not a fan of the script install because as you can see, a simple feature such as faxing has created headaches. I didn't think it would be this difficult to turn on fax to email after a default install out of the box. My Asterisk box has given me nothing but headaches when it comes to fax signals so I figured I would give FS a try in that aspect. Once I change the permissions, I'll reboot the thing just to make sure and then run another test. If this works, I am going to completely rebuild this thing and hopefully the folder creation problem goes away as well. When I get to that point, what source/guide do you guys recommend for a manual install? Robert From: bdfoster at endigotech.com Date: Mon, 21 Jan 2013 14:27:37 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Why are you running Freeswitch as root? Sent from my iPhone On Jan 21, 2013, at 12:59 AM, Robert Craig wrote: As far as I know, FS is running as root. Below are the permissions. <13>:ls -ld storage/fax/3150/inbox drw-rw-rw- 2 root root 4096 Jan 18 10:28 storage/fax/3150/inbox I've gone from the 'freeswitch' folder all the way into the inbox folder verifying the same permissions down the line. Should it include 'execute' permissions as well? Whats even more interesting is that when I create a new fax extension, it doesn't create the folder, such as 3150. I have to manually create it. Robert Date: Sun, 20 Jan 2013 17:07:50 -0300 From: jmesquita at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Robert, you clearly still have a permissions problem on this specific folder (according to your dialplan):/usr/local/freeswitch/storage/fax/3150/inbox/ What is the user FreeSWITCH is running under and what are the current permissions on the folder? Jo?o Mesquita FreeSWITCH? Solutions On Sun, Jan 20, 2013 at 3:36 PM, Robert Craig wrote: George, I'm having a hard time understanding your dialplan. I'm not a FS guru so my apologies. It looks like your dialplan is meant for an ATA. I am just trying to get the email portion configured first; specifically inbound. Once I resolve the inbound issues, I'll work on outbound and then move to ATA's. Robert Date: Sun, 20 Jan 2013 09:52:11 -0500 From: gassaad at emassembly.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Here is mine: Inbound:http://pastebin.freeswitch.org/20471Outbound:http://pastebin.freeswitch.org/20470 Thanks,George On Sat, Jan 19, 2013 at 2:54 PM, Robert Craig wrote: OK, I copied everything from the fax extension dial plan into the below pastebin. http://pastebin.freeswitch.org/20466 Robert From: bdfoster at endigotech.com Date: Fri, 18 Jan 2013 12:59:38 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Submit the dialplan you are using to receive the fax. You can use the FreeSWITCH pastebin, http://pastebin.freeswitch.org Sent from my iPhone On Jan 18, 2013, at 12:43 PM, Robert Craig wrote: I believe I'm just using the built in SpanDSP for faxing. I've looked around for 'rxfax' and can't find anything. I did notice that the directory where the faxes were supposed to be stored for the extension I create didn't even exist. I created them, applied the permissions, but it seems like I am missing something because the same error is still there. Robert Date: Fri, 18 Jan 2013 09:13:36 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration If you are using the rxfax app then you specify the file name and location: Make sure that the target directory has read/write permissions. You can use something like "chmod 666 /my/directory" to make it read/write. -MC On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: Where does the file get downloaded to? What should the permissions be? Robert > Date: Sat, 19 Jan 2013 00:18:45 +0800 > From: steveu at coppice.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If the TIFF file cannot be opened on receive you probably have a file > permissions issue (e.g. with the directory in which the file is to be > created). If the TIFF file cannot be opened on transmit there are > various possibilities about the suitability of the file's contents. > > Steve > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > This was an incoming fax. I don't have much control over the other > > end. Why is that error appearing when trying to receive something? > > > > Robert > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > wrote: > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > >> Please try pdf, you should be in business now. > >> Please let us know, > >> > >> On Jan 18, 2013 12:35 AM, "Robert Craig" >> > wrote: > >> > >> OK, so I've made a little bit of progress. The distant end is a > >> Windows Fax server hooked up to an analog line. I created a fax > >> extension in FS, assigned an inbound route to the extension, and > >> made sure the extension had an email address. I believe the fax > >> receive wants to complete, but there is an interesting error in > >> the log about TIF files. Anyone ever seen this before? > >> > >> v=0 > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> c=IN IP4 72.201.112.131 > >> t=0 0 > >> m=image 27714 udptl t38 > >> a=T38FaxVersion:0 > >> a=T38MaxBitRate:9600 > >> a=T38FaxFillBitRemoval > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxUdpEC:t38UDPRedundancy > >> > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > >> ip/port confirmed. > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > >> Negotiation Result > >> ======================================================= > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > >> status off > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > >> country: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > >> vendor: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > >> model: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > >> processing not successful - result (41) TIFF/F file cannot be > >> opened.* > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > >> transferred: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > >> fax pages: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > >> resolution: 0x0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > >> status off > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > >> country: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > >> vendor: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > >> model: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > >> sofia/external/+14792460547 @flowroute.com > >> Restore previous codec PCMU:0. > >> EXECUTE sofia/external/+14792460547 > >> @flowroute.com hangup() > >> > >> Robert > >> > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> OK, I am not extremely concerned with getting an ATA working > >> right now. If I could just get an inbound extension-to-email > >> working it would be great. I'll start from scratch and then let > >> you know what happens. Just to make sure, I am going to create an > >> extension, assign an email address to it, make sure there is an > >> inbound route to that extension, and then test it. > >> > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > >> From: gassaad at emassembly.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> The reason for outbound extension is: > >> When you send a fax from ATA to the outside world, it will dial > >> first to the outbound extension and then after successful receive > >> it will dial out to the intended number. > >> > >> As for the error you have it may be related to your ATA. I am not > >> sure please see why do you get the following message. > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > >> > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > >> > wrote: > >> > >> OK, I've already tried creating one extension for inbound and > >> set an email address. Of course, it failed and the errors are > >> at the bottom of this email. What is the purpose of the > >> outbound extension? > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > >> From: gassaad at emassembly.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Sorry for the delay to get back to you. > >> 1- App --> fax server > >> 2- Create two extensions (one for inbound) another for > >> (outbound) From within either extension you can send faxes > >> "virtual fax" > >> 3- point your create your outbound and inbound dialplans as > >> normal but make it to point to those extensions. > >> > >> Good luck, > >> > >> George > >> > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > >> > > >> wrote: > >> > >> What did you do on the config side to get it to work? > >> I've tried both an Ata and a fax to email extension and > >> they both fail in both directions. > >> > >> Robert > >> > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > >> > > >> wrote: > >> > >> I use Fusion successfully to send / receive fax. > >> The way it works: > >> ATA --->FaxServer. > >> After successful receipt, Fax Server---> DID where > >> you have dialled originally. > >> > >> Cheers, > >> > >> George > >> > >> > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > >> >> > wrote: > >> > >> Did anyone see the issue or what I may have > >> missed in the config? As mentioned earlier, I > >> thought it would after a fresh install. Any help > >> is appreciated. > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Thanks for the quick response. Below is the > >> capture from the log viewer. In this test, I am > >> using a remote fax machine (line from Cox with > >> physical machine) faxing into the "email server" > >> where I created an extension for fax-to-email. > >> Calling number is 1479-246-0547. > >> > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:99 > >> sofia/external/+14792460547 at flowroute.com > >> Standard DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> mod_sofia.c:396 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:559 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State Change > >> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:556 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate Change HANGUP > >> -> DOWN > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1510 Close Channel > >> sofia/external/+14792460547 at flowroute.com > >> [CS_DESTROY] > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1506 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Ended > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1488 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Locked, Waiting on > >> external entities > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:440 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change CS_REPORTING > >> -> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State > >> REPORTING going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:92 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> REPORTING, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:446 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_HANGUP -> CS_REPORTING > >> > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:599 Hangup Command > >> with no Session system(/usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 > >> name=+14792460547-2013-01-10-13-39-26 > >> messages='result: Disconnected after permitted > >> retries sender:ATW-4792460547 pages:0' > >> domain=192.168.4.15 caller_id_name='' > >> caller_id_number=): > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:48 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> HANGUP, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:552 Sending BYE to > >> sofia/external/+14792460547 at flowroute.com > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:503 Channel > >> sofia/external/+14792460547 at flowroute.com > >> hanging up, > >> cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:2687 > >> sofia/external/+14792460547 at flowroute.com > >> skip receive > >> message [APPLICATION_EXEC_COMPLETE] (channel is > >> hungup already) > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:3017 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [KILL] > >> 2013-01-10 13:39:52.216151 [NOTICE] > >> mod_dptools.c:1150 Hangup > >> sofia/external/+14792460547 at flowroute.com > >> [CS_EXECUTE] > >> [NORMAL_CLEARING] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:2994 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change ACTIVE -> HANGUP > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> hangup() > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_codec.c:244 > >> sofia/external/+14792460547 at flowroute.com > >> Restore > >> previous codec PCMU:0. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:516 remote model: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:515 remote vendor: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:514 remote country: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:513 ECM status off > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:509 Total fax pages: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:507 Pages transferred: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > >> Fax Ident > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:505 Remote station id: > >> ATW-4792460547 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:500 Fax processing not > >> successful - result (48) Disconnected after > >> permitted retries. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> 2013-01-10 13:39:38.766190 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:29.576127 [ERR] > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > >> must be between 20 and 10000 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxRateManagement:transferredTCF > >> m=image 31632 udptl t38 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completing][200] > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > >> sofia/external/+14792460547 at flowroute.com > >> Update Callee > >> ID to "+14792460547" <+14792460547> > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [calling][0] > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:2647 Local SDP: > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1379 Raw write codec activation > >> Success L16 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_core_codec.c:219 > >> sofia/external/+14792460547 at flowroute.com > >> Push codec L16:70 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1363 Raw read codec activation > >> Success L16 20000 > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > >> > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_ivr_play_say.c:1682 done playing file > >> silence_stream://2000 > >> 2013-01-10 13:39:26.256197 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> switch_ivr_play_say.c:1309 Codec Activated > >> L16 at 8000hz 1 channels 20ms > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> playback(silence_stream://2000) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38_request]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38_request=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38]=[true] > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completed][200] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.006127 [NOTICE] > >> mod_dptools.c:1176 Channel > >> [sofia/external/+14792460547 at flowroute.com > >> ] has been > >> answered > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_channel.c:3395 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change RINGING -> ACTIVE > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=sendrecv > >> a=ptime:20 > >> a=silenceSupp:off - - - - > >> a=fmtp:101 0-16 > >> a=rtpmap:101 telephone-event/8000 > >> a=rtpmap:0 PCMU/8000 > >> m=audio 27588 RTP/AVP 0 101 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_sofia.c:856 Local SDP > >> sofia/external/+14792460547 at flowroute.com > >> : > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:1046 STUN Success > >> [72.201.112.131]:[27588] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> answer() > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [api_hangup_hook]=[system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=${last_fax} messages='result: > >> ${fax_result_text} > >> sender:${fax_remote_station_id} > >> pages:${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='${caller_id_name}' > >> caller_id_number=${caller_id_number}] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\${last_fax} > >> messages='result: \${fax_result_text} > >> sender:\${fax_remote_station_id} > >> pages:\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\${caller_id_name}' > >> caller_id_number=\${caller_id_number}) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [call_direction]=[local] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(call_direction=local) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action hangup() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> playback(silence_stream://2000) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38_request=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action answer() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\\\${last_fax} > >> messages='result: \\\${fax_result_text} > >> sender:\\\${fax_remote_station_id} > >> pages:\\\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\\\${caller_id_name}' > >> caller_id_number=\\\${caller_id_number}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test Fax] destination_number(3150) =~ > >> /^3150$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Test Fax] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain_user] destination_number(3150) =~ > >> /^\*97$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain_user] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain] destination_number(3150) =~ > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [send_to_voicemail] > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->send_to_voicemail] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [extension-intercom] > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->extension-intercom] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [intercept-ext] destination_number(3150) > >> =~ /^\*\*(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->intercept-ext] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_return] destination_number(3150) =~ > >> /^\*69$|^lcr$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_return] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_privacy] destination_number(3150) =~ > >> /^\*67(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_privacy] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [global] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > >> break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->global] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [redial] destination_number(3150) =~ > >> /^(redial|\*870)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->redial] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [group-intercept] destination_number(3150) > >> =~ /^\*8$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->group-intercept] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> ANTI-Action > >> set(call_direction=local) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call-direction] ${call_direction}() =~ > >> /^(inbound|outbound|local)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call-direction] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [Flowroute.11d] destination_number(3150) > >> =~ /^\+?(\d{11})$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Flowroute.11d] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->3150 in context default > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:25.816280 [NOTICE] > >> switch_ivr.c:1781 Transfer > >> sofia/external/+14792460547 at flowroute.com > >> to > >> XML[3150 at default] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_ivr.c:1774 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_EXECUTE -> CS_ROUTING > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> transfer(3150 > >> XML default) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_channel.c:1135 EXPORT (export_vars) > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [outside_call]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(outside_call=true) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> transfer(3150 XML default) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] > >> destination_number(15203039810) =~ > >> /^15203039810$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] context(public) =~ /public/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->Test_Number] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_debug] ${call_debug}(false) =~ > >> /^true$/ break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->call_debug] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(outside_call=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [outside_call] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->outside_call] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [public->unloop] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->15203039810 in context public > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> References: <50D8D157.7020409@orresta.no-ip.com> <50FDB180.3010404@orresta.no-ip.com> Message-ID: Lets start with patches for all the code changes on jira. We also will need a full set of phrase macros for swedish (again patches). For the sound files we typically start with 48k and resample down, for those coordinate with Michael Collins. Mike On Jan 21, 2013, at 4:22 PM, Jakob wrote: > Ok, > > I now have a full set of recordings for mod_say_sv and mod_voicemail. In what format should I submit them? 48k.wav? 32? 16? ulaw? and to where? > > I also have a working mod_say_sv module written with the english example as a base. So it is aware of the different encodings and not hardcoded to .wav files. This is now in production on my box and have been solid. > > There are however a patch to the FS-core that were needed. Adding a new gender to make the natural flow of recordings working. And a few patches to build files. > > I'm still working on getting the full voicemail system all swedish. > > How can I submit this for the benefit of others? > > Regards > Jakob Sundberg > > 12/26/12 19:41, Michael Collins skrev: >> Also, if you get a set of Swedish recordings let us know and we'll get them hosted on our files.freeswitch.org server. >> -MC >> >> On Mon, Dec 24, 2012 at 2:17 PM, Avi Marcus wrote: >> I can't find a list on the wiki.. but there doesn't seem to be a mod_say_** for swedish prompts. >> see in git source: src/mod/say >> >> If one of the languages is similar in making things singular/plural, masculine/femanine you can use that as a template. >> >> Or you can create speech phrases for numbers with the logic in there, but that's a hack.. >> >> >> -Avi >> >> >> On Tue, Dec 25, 2012 at 12:04 AM, Jakob wrote: >> Hi, >> >> I started working on a project that would include a voicemail app. but I >> have trouble finding a swedish implementation of a mod_say so that I can >> populate it with swedish prompts. >> >> I already have an working IVR, in swedish but there are still a few >> bugs, mainly getting english numbers in my swedish prompts. >> >> Jakob >> >> _________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/a3987a99/attachment.html From william.king at quentustech.com Tue Jan 22 00:47:34 2013 From: william.king at quentustech.com (William King) Date: Mon, 21 Jan 2013 13:47:34 -0800 Subject: [Freeswitch-users] Reload mod_lua In-Reply-To: References: <50FD7E4F.6080607@gmail.com> <50FDAFAE.8080608@gmail.com> Message-ID: <50FDB776.3060100@quentustech.com> Interesting. A mod_lua api that would list the currently running lua scripts with an id that could be used to kill off individual scripts. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 01/21/2013 01:21 PM, Avi Marcus wrote: > A tool that can monitor Lua threads, and kill them, would come in useful > every so often... > > -Avi > > On Mon, Jan 21, 2013 at 11:14 PM, Abaci > wrote: > > or you can have your script listen for event or check a variable and > exit the script as needed. and another background script which will > start the script again if it's not running. not a simple solution > but doable. > > > On 1/21/2013 1:52 PM, Michael Collins wrote: >> Lua is not "unloadable" and therefore you cannot do this the way >> you are trying to. You'll need to restart FreeSWITCH. >> -MC >> >> On Mon, Jan 21, 2013 at 9:43 AM, Mimiko > > wrote: >> >> I want to reload mod_lua in order to reload lua scripts that a >> starte >> automatically on start-up. But I get this: >> >> freeswitch at internal> reload mod_lua >> +OK Reloading XML >> -ERR unloading module [Module is not unloadable] >> -ERR loading module [Module already loaded] >> >> [CRIT] switch_loadable_module.c:1451 Module is not unloadable. >> [WARNING] switch_loadable_module.c:1400 Module mod_lua Already >> Loaded! >> [INFO] mod_enum.c:872 ENUM Reloaded >> [INFO] switch_time.c:1165 Timezone reloaded 530 definitions >> >> How to refresh modified lua scripts without reloading FS with >> restart? >> -- >> Mimiko desu. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Tue Jan 22 01:02:38 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 14:02:38 -0800 Subject: [Freeswitch-users] anybody working on a swedish mod_say? In-Reply-To: <50FDB180.3010404@orresta.no-ip.com> References: <50D8D157.7020409@orresta.no-ip.com> <50FDB180.3010404@orresta.no-ip.com> Message-ID: Please open a jira at jira.freeswitch.org and add the patch(es) to the new ticket. Add it to the main FS project. Also, be sure that your patches follow these guidelines: http://wiki.freeswitch.org/wiki/Submitting_Patches Thanks! -MC On Mon, Jan 21, 2013 at 1:22 PM, Jakob wrote: > Ok, > > I now have a full set of recordings for mod_say_sv and mod_voicemail. In > what format should I submit them? 48k.wav? 32? 16? ulaw? and to where? > > I also have a working mod_say_sv module written with the english example > as a base. So it is aware of the different encodings and not hardcoded to > .wav files. This is now in production on my box and have been solid. > > There are however a patch to the FS-core that were needed. Adding a new > gender to make the natural flow of recordings working. And a few patches to > build files. > > I'm still working on getting the full voicemail system all swedish. > > How can I submit this for the benefit of others? > > Regards > Jakob Sundberg > > 12/26/12 19:41, Michael Collins skrev: > > Also, if you get a set of Swedish recordings let us know and we'll get > them hosted on our files.freeswitch.org server. > -MC > > On Mon, Dec 24, 2012 at 2:17 PM, Avi Marcus wrote: > >> I can't find a list on the wiki.. but there doesn't seem to be a >> mod_say_** for swedish prompts. >> see in git source: src/mod/say >> >> If one of the languages is similar in making things singular/plural, >> masculine/femanine you can use that as a template. >> >> Or you can create speech phrases for numbers with the logic in there, >> but that's a hack.. >> >> >> -Avi >> >> >> On Tue, Dec 25, 2012 at 12:04 AM, Jakob wrote: >> >>> Hi, >>> >>> I started working on a project that would include a voicemail app. but I >>> have trouble finding a swedish implementation of a mod_say so that I can >>> populate it with swedish prompts. >>> >>> I already have an working IVR, in swedish but there are still a few >>> bugs, mainly getting english numbers in my swedish prompts. >>> >>> Jakob >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/e69a4924/attachment-0001.html From msc at freeswitch.org Tue Jan 22 01:03:47 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 14:03:47 -0800 Subject: [Freeswitch-users] anybody working on a swedish mod_say? In-Reply-To: References: <50D8D157.7020409@orresta.no-ip.com> <50FDB180.3010404@orresta.no-ip.com> Message-ID: Indeed. Please email me directly with the location where I can download the 48k version of your sound set. I'll take it from there. -MC On Mon, Jan 21, 2013 at 1:41 PM, Michael Jerris wrote: > Lets start with patches for all the code changes on jira. We also will > need a full set of phrase macros for swedish (again patches). For the > sound files we typically start with 48k and resample down, for those > coordinate with Michael Collins. > > Mike > > On Jan 21, 2013, at 4:22 PM, Jakob wrote: > > Ok, > > I now have a full set of recordings for mod_say_sv and mod_voicemail. In > what format should I submit them? 48k.wav? 32? 16? ulaw? and to where? > > I also have a working mod_say_sv module written with the english example > as a base. So it is aware of the different encodings and not hardcoded to > .wav files. This is now in production on my box and have been solid. > > There are however a patch to the FS-core that were needed. Adding a new > gender to make the natural flow of recordings working. And a few patches to > build files. > > I'm still working on getting the full voicemail system all swedish. > > How can I submit this for the benefit of others? > > Regards > Jakob Sundberg > > 12/26/12 19:41, Michael Collins skrev: > > Also, if you get a set of Swedish recordings let us know and we'll get > them hosted on our files.freeswitch.org server. > -MC > > On Mon, Dec 24, 2012 at 2:17 PM, Avi Marcus wrote: > >> I can't find a list on the wiki.. but there doesn't seem to be a >> mod_say_** for swedish prompts. >> see in git source: src/mod/say >> >> If one of the languages is similar in making things singular/plural, >> masculine/femanine you can use that as a template. >> >> Or you can create speech phrases for numbers with the logic in there, >> but that's a hack.. >> >> >> -Avi >> >> >> On Tue, Dec 25, 2012 at 12:04 AM, Jakob wrote: >> >>> Hi, >>> >>> I started working on a project that would include a voicemail app. but I >>> have trouble finding a swedish implementation of a mod_say so that I can >>> populate it with swedish prompts. >>> >>> I already have an working IVR, in swedish but there are still a few >>> bugs, mainly getting english numbers in my swedish prompts. >>> >>> Jakob >>> >>> _________ >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/d6a5d6c8/attachment.html From msc at freeswitch.org Tue Jan 22 01:07:15 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 14:07:15 -0800 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: <50F975E5.2040206@coppice.org> <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com> Message-ID: I'd probably start here: http://wiki.freeswitch.org/wiki/Mod_voicemail#Debugging_an_external_mailer -MC On Mon, Jan 21, 2013 at 1:39 PM, Robert Craig wrote: > It worked! The read/write/execute permissions fixed it. Now, oddly enough, > the PDF didn't get emailed out. I have an email address assigned to the fax > extension, but it apparently never sent out. I don't see anything in the > logs. Can someone point me in the right direction on making it manually > send, or configured the email server? Thanks. > > > Robert > > > ------------------------------ > From: craig.robert at hotmail.com > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 21 Jan 2013 14:10:52 -0700 > Subject: Re: [Freeswitch-users] T38 Configuration > > OK, so I'll change the permissions to include read/write/execute. FS is > running as root because this is a box at the house I am playing with. I > installed FS using one of the script methods out there and I was logged in > as root at the time when I ran everything. Regardless, I'm not a fan of the > script install because as you can see, a simple feature such as faxing has > created headaches. I didn't think it would be this difficult to turn on fax > to email after a default install out of the box. My Asterisk box has given > me nothing but headaches when it comes to fax signals so I figured I would > give FS a try in that aspect. Once I change the permissions, I'll reboot > the thing just to make sure and then run another test. If this works, I am > going to completely rebuild this thing and hopefully the folder creation > problem goes away as well. When I get to that point, what source/guide do > you guys recommend for a manual install? > > > Robert > > > ------------------------------ > From: bdfoster at endigotech.com > Date: Mon, 21 Jan 2013 14:27:37 -0500 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Why are you running Freeswitch as root? > > Sent from my iPhone > > On Jan 21, 2013, at 12:59 AM, Robert Craig > wrote: > > As far as I know, FS is running as root. Below are the permissions. > > <13>:ls -ld storage/fax/3150/inbox > > drw-rw-rw- 2 root root 4096 Jan 18 10:28 storage/fax/3150/inbox > > I've gone from the 'freeswitch' folder all the way into the inbox folder > verifying the same permissions down the line. Should it include 'execute' > permissions as well? Whats even more interesting is that when I create a > new fax extension, it doesn't create the folder, such as 3150. I have to > manually create it. > > > > Robert > > > ------------------------------ > Date: Sun, 20 Jan 2013 17:07:50 -0300 > From: jmesquita at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Robert, you clearly still have a permissions problem on this specific > folder (according to your dialplan): > /usr/local/freeswitch/storage/fax/3150/inbox/ > > What is the user FreeSWITCH is running under and what are the current > permissions on the folder? > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Sun, Jan 20, 2013 at 3:36 PM, Robert Craig wrote: > > George, > > I'm having a hard time understanding your dialplan. I'm not a FS guru > so my apologies. It looks like your dialplan is meant for an ATA. I am just > trying to get the email portion configured first; specifically inbound. > Once I resolve the inbound issues, I'll work on outbound and then move to > ATA's. > > > Robert > > > ------------------------------ > Date: Sun, 20 Jan 2013 09:52:11 -0500 > From: gassaad at emassembly.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Here is mine: > Inbound: > http://pastebin.freeswitch.org/20471 > Outbound: > http://pastebin.freeswitch.org/20470 > > Thanks, > George > > On Sat, Jan 19, 2013 at 2:54 PM, Robert Craig wrote: > > OK, I copied everything from the fax extension dial plan into the below > pastebin. > > http://pastebin.freeswitch.org/20466 > > > Robert > > > ------------------------------ > From: bdfoster at endigotech.com > Date: Fri, 18 Jan 2013 12:59:38 -0500 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Submit the dialplan you are using to receive the fax. You can use the > FreeSWITCH pastebin, http://pastebin.freeswitch.org > > Sent from my iPhone > > On Jan 18, 2013, at 12:43 PM, Robert Craig > wrote: > > I believe I'm just using the built in SpanDSP for faxing. I've looked > around for 'rxfax' and can't find anything. I did notice that the directory > where the faxes were supposed to be stored for the extension I create > didn't even exist. I created them, applied the permissions, but it seems > like I am missing something because the same error is still there. > > > Robert > > > ------------------------------ > Date: Fri, 18 Jan 2013 09:13:36 -0800 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If you are using the rxfax app then you specify the file name and location: > > > > Make sure that the target directory has read/write permissions. You can > use something like "chmod 666 /my/directory" to make it read/write. > > -MC > > On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: > > Where does the file get downloaded to? What should the permissions be? > > Robert > > > > Date: Sat, 19 Jan 2013 00:18:45 +0800 > > From: steveu at coppice.org > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] T38 Configuration > > > > If the TIFF file cannot be opened on receive you probably have a file > > permissions issue (e.g. with the directory in which the file is to be > > created). If the TIFF file cannot be opened on transmit there are > > various possibilities about the suitability of the file's contents. > > > > Steve > > > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > > This was an incoming fax. I don't have much control over the other > > > end. Why is that error appearing when trying to receive something? > > > > > > Robert > > > > > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > > wrote: > > > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > > >> Please try pdf, you should be in business now. > > >> Please let us know, > > >> > > >> On Jan 18, 2013 12:35 AM, "Robert Craig" > >> > wrote: > > >> > > >> OK, so I've made a little bit of progress. The distant end is a > > >> Windows Fax server hooked up to an analog line. I created a fax > > >> extension in FS, assigned an inbound route to the extension, and > > >> made sure the extension had an email address. I believe the fax > > >> receive wants to complete, but there is an interesting error in > > >> the log about TIF files. Anyone ever seen this before? > > >> > > >> v=0 > > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> c=IN IP4 72.201.112.131 > > >> t=0 0 > > >> m=image 27714 udptl t38 > > >> a=T38FaxVersion:0 > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxFillBitRemoval > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > > >> ip/port confirmed. > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > > >> Negotiation Result > > >> ======================================================= > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > > >> status off > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > > >> country: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > > >> vendor: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > > >> model: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > > >> processing not successful - result (41) TIFF/F file cannot be > > >> opened.* > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > > >> transferred: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > > >> fax pages: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > > >> resolution: 0x0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > > >> status off > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > > >> country: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > > >> vendor: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > > >> model: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > > >> sofia/external/+14792460547 @flowroute.com > > >> Restore previous codec PCMU:0. > > >> EXECUTE sofia/external/+14792460547 > > >> @flowroute.com hangup() > > >> > > >> Robert > > >> > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> OK, I am not extremely concerned with getting an ATA working > > >> right now. If I could just get an inbound extension-to-email > > >> working it would be great. I'll start from scratch and then let > > >> you know what happens. Just to make sure, I am going to create an > > >> extension, assign an email address to it, make sure there is an > > >> inbound route to that extension, and then test it. > > >> > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > > >> From: gassaad at emassembly.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> The reason for outbound extension is: > > >> When you send a fax from ATA to the outside world, it will dial > > >> first to the outbound extension and then after successful receive > > >> it will dial out to the intended number. > > >> > > >> As for the error you have it may be related to your ATA. I am not > > >> sure please see why do you get the following message. > > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > > >> > > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > > >> > wrote: > > >> > > >> OK, I've already tried creating one extension for inbound and > > >> set an email address. Of course, it failed and the errors are > > >> at the bottom of this email. What is the purpose of the > > >> outbound extension? > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > > >> From: gassaad at emassembly.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Sorry for the delay to get back to you. > > >> 1- App --> fax server > > >> 2- Create two extensions (one for inbound) another for > > >> (outbound) From within either extension you can send faxes > > >> "virtual fax" > > >> 3- point your create your outbound and inbound dialplans as > > >> normal but make it to point to those extensions. > > >> > > >> Good luck, > > >> > > >> George > > >> > > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > > >> > > > >> wrote: > > >> > > >> What did you do on the config side to get it to work? > > >> I've tried both an Ata and a fax to email extension and > > >> they both fail in both directions. > > >> > > >> Robert > > >> > > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > > >> > > > >> wrote: > > >> > > >> I use Fusion successfully to send / receive fax. > > >> The way it works: > > >> ATA --->FaxServer. > > >> After successful receipt, Fax Server---> DID where > > >> you have dialled originally. > > >> > > >> Cheers, > > >> > > >> George > > >> > > >> > > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > > >> > >> > wrote: > > >> > > >> Did anyone see the issue or what I may have > > >> missed in the config? As mentioned earlier, I > > >> thought it would after a fresh install. Any help > > >> is appreciated. > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Thanks for the quick response. Below is the > > >> capture from the log viewer. In this test, I am > > >> using a remote fax machine (line from Cox with > > >> physical machine) faxing into the "email server" > > >> where I created an extension for fax-to-email. > > >> Calling number is 1479-246-0547. > > >> > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:99 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> mod_sofia.c:396 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:559 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State Change > > >> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:556 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate Change HANGUP > > >> -> DOWN > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1510 Close Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_DESTROY] > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1506 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Ended > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1488 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Locked, Waiting on > > >> external entities > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:440 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change CS_REPORTING > > >> -> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State > > >> REPORTING going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:92 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> REPORTING, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:446 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_HANGUP -> CS_REPORTING > > >> > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:599 Hangup Command > > >> with no Session system(/usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 > > >> name=+14792460547-2013-01-10-13-39-26 > > >> messages='result: Disconnected after permitted > > >> retries sender:ATW-4792460547 pages:0' > > >> domain=192.168.4.15 caller_id_name='' > > >> caller_id_number=): > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:48 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> HANGUP, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:552 Sending BYE to > > >> sofia/external/+14792460547 at flowroute.com > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:503 Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> hanging up, > > >> cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:2687 > > >> sofia/external/+14792460547 at flowroute.com > > >> skip receive > > >> message [APPLICATION_EXEC_COMPLETE] (channel is > > >> hungup already) > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:3017 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [KILL] > > >> 2013-01-10 13:39:52.216151 [NOTICE] > > >> mod_dptools.c:1150 Hangup > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_EXECUTE] > > >> [NORMAL_CLEARING] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:2994 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change ACTIVE -> HANGUP > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> hangup() > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_codec.c:244 > > >> sofia/external/+14792460547 at flowroute.com > > >> Restore > > >> previous codec PCMU:0. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:516 remote model: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:515 remote vendor: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:514 remote country: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:513 ECM status off > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:509 Total fax pages: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:507 Pages transferred: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > > >> Fax Ident > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:505 Remote station id: > > >> ATW-4792460547 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:500 Fax processing not > > >> successful - result (48) Disconnected after > > >> permitted retries. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:38.766190 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:29.576127 [ERR] > > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > > >> must be between 20 and 10000 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxRateManagement:transferredTCF > > >> m=image 31632 udptl t38 > > >> t=0 0 > > >> c=IN IP4 66.42.121.5 > > >> s=- > > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > > >> v=0 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > > >> Remote SDP: > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completing][200] > > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > > >> sofia/external/+14792460547 at flowroute.com > > >> Update Callee > > >> ID to "+14792460547" <+14792460547> > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [calling][0] > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:2647 Local SDP: > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1379 Raw write codec activation > > >> Success L16 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_core_codec.c:219 > > >> sofia/external/+14792460547 at flowroute.com > > >> Push codec L16:70 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1363 Raw read codec activation > > >> Success L16 20000 > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > > >> > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_ivr_play_say.c:1682 done playing file > > >> silence_stream://2000 > > >> 2013-01-10 13:39:26.256197 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> switch_ivr_play_say.c:1309 Codec Activated > > >> L16 at 8000hz 1 channels 20ms > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> playback(silence_stream://2000) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38_request]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38_request=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38]=[true] > > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completed][200] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.006127 [NOTICE] > > >> mod_dptools.c:1176 Channel > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] has been > > >> answered > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_channel.c:3395 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change RINGING -> ACTIVE > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=sendrecv > > >> a=ptime:20 > > >> a=silenceSupp:off - - - - > > >> a=fmtp:101 0-16 > > >> a=rtpmap:101 telephone-event/8000 > > >> a=rtpmap:0 PCMU/8000 > > >> m=audio 27588 RTP/AVP 0 101 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_sofia.c:856 Local SDP > > >> sofia/external/+14792460547 at flowroute.com > > >> : > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:1046 STUN Success > > >> [72.201.112.131]:[27588] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> answer() > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [api_hangup_hook]=[system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=${last_fax} messages='result: > > >> ${fax_result_text} > > >> sender:${fax_remote_station_id} > > >> pages:${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='${caller_id_name}' > > >> caller_id_number=${caller_id_number}] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\${last_fax} > > >> messages='result: \${fax_result_text} > > >> sender:\${fax_remote_station_id} > > >> pages:\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\${caller_id_name}' > > >> caller_id_number=\${caller_id_number}) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-spymap/+14792460547 > /d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [call_direction]=[local] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(call_direction=local) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action hangup() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> playback(silence_stream://2000) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38_request=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action answer() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\\\${last_fax} > > >> messages='result: \\\${fax_result_text} > > >> sender:\\\${fax_remote_station_id} > > >> pages:\\\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\\\${caller_id_name}' > > >> caller_id_number=\\\${caller_id_number}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test Fax] destination_number(3150) =~ > > >> /^3150$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Test Fax] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain_user] destination_number(3150) =~ > > >> /^\*97$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain_user] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain] destination_number(3150) =~ > > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [send_to_voicemail] > > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->send_to_voicemail] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [extension-intercom] > > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->extension-intercom] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [intercept-ext] destination_number(3150) > > >> =~ /^\*\*(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->intercept-ext] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_return] destination_number(3150) =~ > > >> /^\*69$|^lcr$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_return] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_privacy] destination_number(3150) =~ > > >> /^\*67(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_privacy] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [global] > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > > >> break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->global] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [redial] destination_number(3150) =~ > > >> /^(redial|\*870)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->redial] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [group-intercept] destination_number(3150) > > >> =~ /^\*8$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->group-intercept] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> ANTI-Action > > >> set(call_direction=local) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call-direction] ${call_direction}() =~ > > >> /^(inbound|outbound|local)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call-direction] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [Flowroute.11d] destination_number(3150) > > >> =~ /^\+?(\d{11})$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Flowroute.11d] continue=false > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->3150 in context default > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [NOTICE] > > >> switch_ivr.c:1781 Transfer > > >> sofia/external/+14792460547 at flowroute.com > > >> to > > >> XML[3150 at default] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_ivr.c:1774 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_EXECUTE -> CS_ROUTING > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> transfer(3150 > > >> XML default) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_channel.c:1135 EXPORT (export_vars) > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [outside_call]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(outside_call=true) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> transfer(3150 XML default) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] > > >> destination_number(15203039810) =~ > > >> /^15203039810$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] context(public) =~ /public/ > > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->Test_Number] continue=false > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_debug] ${call_debug}(false) =~ > > > >> /^true$/ break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->call_debug] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(outside_call=true) > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [outside_call] > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->outside_call] continue=true > > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > > >> Regex > > >> (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [public->unloop] continue=false > > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->15203039810 in context public > > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> > ... > > [Message clipped] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http:// > lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/9d1d32fc/attachment-0001.html From avi at avimarcus.net Tue Jan 22 01:09:34 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 22 Jan 2013 00:09:34 +0200 Subject: [Freeswitch-users] anybody working on a swedish mod_say? In-Reply-To: <50FDB180.3010404@orresta.no-ip.com> References: <50D8D157.7020409@orresta.no-ip.com> <50FDB180.3010404@orresta.no-ip.com> Message-ID: There are other mod_say languages that have genders. Yours requires a patch to the core? -Avi On Mon, Jan 21, 2013 at 11:22 PM, Jakob wrote: > Ok, > > I now have a full set of recordings for mod_say_sv and mod_voicemail. In > what format should I submit them? 48k.wav? 32? 16? ulaw? and to where? > > I also have a working mod_say_sv module written with the english example > as a base. So it is aware of the different encodings and not hardcoded to > .wav files. This is now in production on my box and have been solid. > > There are however a patch to the FS-core that were needed. Adding a new > gender to make the natural flow of recordings working. And a few patches to > build files. > > I'm still working on getting the full voicemail system all swedish. > > How can I submit this for the benefit of others? > > Regards > Jakob Sundberg > > 12/26/12 19:41, Michael Collins skrev: > > Also, if you get a set of Swedish recordings let us know and we'll get > them hosted on our files.freeswitch.org server. > -MC > > On Mon, Dec 24, 2012 at 2:17 PM, Avi Marcus wrote: > >> I can't find a list on the wiki.. but there doesn't seem to be a >> mod_say_** for swedish prompts. >> see in git source: src/mod/say >> >> If one of the languages is similar in making things singular/plural, >> masculine/femanine you can use that as a template. >> >> Or you can create speech phrases for numbers with the logic in there, >> but that's a hack.. >> >> >> -Avi >> >> >> On Tue, Dec 25, 2012 at 12:04 AM, Jakob wrote: >> >>> Hi, >>> >>> I started working on a project that would include a voicemail app. but I >>> have trouble finding a swedish implementation of a mod_say so that I can >>> populate it with swedish prompts. >>> >>> I already have an working IVR, in swedish but there are still a few >>> bugs, mainly getting english numbers in my swedish prompts. >>> >>> Jakob >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/9c81f9ac/attachment.html From Hector.Geraldino at ipsoft.com Tue Jan 22 01:17:20 2013 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Mon, 21 Jan 2013 22:17:20 +0000 Subject: [Freeswitch-users] Make a call with Perl In-Reply-To: <50FD5C8B.9030306@iway.ch> References: <50FD5C8B.9030306@iway.ch> Message-ID: I'm not a perl expert, but in general I think you need to follow this pattern: 1) Establish the connection to FreeSWITCH, you'll be using the event socket on inbound mode 2) Start listening for the events you want to capture (send the 'event plain all' or 'event plain CHANNEL_ANSWER DTMF CHANNEL_HANGUP ...' command) 3) Create an uuid (by sending the create_uuid command) so you can identify which call you'll be listening events for 4) Add a filter for this uuid, so you'll only receive events for this call over this connection (filter Unique-ID + uuid) 5) originate the call: {origination_uuid= + uuid}sofia/internal/NXX at GATEWAY Check this simple perl example: http://wiki.freeswitch.org/wiki/Event_Socket_Library#Simple_Perl_Example -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matthias Cramer Sent: Monday, January 21, 2013 10:20 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Make a call with Perl Hi All I try to make a Perl Script which calls a certain external Phone number. It then should wait until someone picks up and play a sound file The user on the other end then needs to confirm the call with a pin Then another sound file should be played and the the call should be terminated. I tried several ways with ESL and originate. But I allays end up with a call I have no control over. I can read DTMF but can not hang up the call or play a second file. I think I am going in to a wrong direction. Any hints ? Regards Matthias -- Matthias Cramer / mc322-ripe Senior Network & Security Engineer iway AG Phone +41 43 500 1111 Badenerstrasse 569 Fax +41 44 271 3535 CH-8048 Z?rich http://www.iway.ch/ GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250 From jmesquita at freeswitch.org Tue Jan 22 01:49:40 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 21 Jan 2013 19:49:40 -0300 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: <50F975E5.2040206@coppice.org> <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com> Message-ID: Just a correction.. Faxing is NOT a simple thing. Specially when it is dealing with VoIP systems... Jo?o Mesquita FreeSWITCH? Solutions On Mon, Jan 21, 2013 at 6:10 PM, Robert Craig wrote: > OK, so I'll change the permissions to include read/write/execute. FS is > running as root because this is a box at the house I am playing with. I > installed FS using one of the script methods out there and I was logged in > as root at the time when I ran everything. Regardless, I'm not a fan of the > script install because as you can see, a simple feature such as faxing has > created headaches. I didn't think it would be this difficult to turn on fax > to email after a default install out of the box. My Asterisk box has given > me nothing but headaches when it comes to fax signals so I figured I would > give FS a try in that aspect. Once I change the permissions, I'll reboot > the thing just to make sure and then run another test. If this works, I am > going to completely rebuild this thing and hopefully the folder creation > problem goes away as well. When I get to that point, what source/guide do > you guys recommend for a manual install? > > > Robert > > > ------------------------------ > From: bdfoster at endigotech.com > Date: Mon, 21 Jan 2013 14:27:37 -0500 > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Why are you running Freeswitch as root? > > Sent from my iPhone > > On Jan 21, 2013, at 12:59 AM, Robert Craig > wrote: > > As far as I know, FS is running as root. Below are the permissions. > > <13>:ls -ld storage/fax/3150/inbox > > drw-rw-rw- 2 root root 4096 Jan 18 10:28 storage/fax/3150/inbox > > I've gone from the 'freeswitch' folder all the way into the inbox folder > verifying the same permissions down the line. Should it include 'execute' > permissions as well? Whats even more interesting is that when I create a > new fax extension, it doesn't create the folder, such as 3150. I have to > manually create it. > > > > Robert > > > ------------------------------ > Date: Sun, 20 Jan 2013 17:07:50 -0300 > From: jmesquita at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Robert, you clearly still have a permissions problem on this specific > folder (according to your dialplan): > /usr/local/freeswitch/storage/fax/3150/inbox/ > > What is the user FreeSWITCH is running under and what are the current > permissions on the folder? > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Sun, Jan 20, 2013 at 3:36 PM, Robert Craig wrote: > > George, > > I'm having a hard time understanding your dialplan. I'm not a FS guru > so my apologies. It looks like your dialplan is meant for an ATA. I am just > trying to get the email portion configured first; specifically inbound. > Once I resolve the inbound issues, I'll work on outbound and then move to > ATA's. > > > Robert > > > ------------------------------ > Date: Sun, 20 Jan 2013 09:52:11 -0500 > From: gassaad at emassembly.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Here is mine: > Inbound: > http://pastebin.freeswitch.org/20471 > Outbound: > http://pastebin.freeswitch.org/20470 > > Thanks, > George > > On Sat, Jan 19, 2013 at 2:54 PM, Robert Craig wrote: > > OK, I copied everything from the fax extension dial plan into the below > pastebin. > > http://pastebin.freeswitch.org/20466 > > > Robert > > > ------------------------------ > From: bdfoster at endigotech.com > Date: Fri, 18 Jan 2013 12:59:38 -0500 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > Submit the dialplan you are using to receive the fax. You can use the > FreeSWITCH pastebin, http://pastebin.freeswitch.org > > Sent from my iPhone > > On Jan 18, 2013, at 12:43 PM, Robert Craig > wrote: > > I believe I'm just using the built in SpanDSP for faxing. I've looked > around for 'rxfax' and can't find anything. I did notice that the directory > where the faxes were supposed to be stored for the extension I create > didn't even exist. I created them, applied the permissions, but it seems > like I am missing something because the same error is still there. > > > Robert > > > ------------------------------ > Date: Fri, 18 Jan 2013 09:13:36 -0800 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If you are using the rxfax app then you specify the file name and location: > > > > Make sure that the target directory has read/write permissions. You can > use something like "chmod 666 /my/directory" to make it read/write. > > -MC > > On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: > > Where does the file get downloaded to? What should the permissions be? > > Robert > > > > Date: Sat, 19 Jan 2013 00:18:45 +0800 > > From: steveu at coppice.org > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] T38 Configuration > > > > If the TIFF file cannot be opened on receive you probably have a file > > permissions issue (e.g. with the directory in which the file is to be > > created). If the TIFF file cannot be opened on transmit there are > > various possibilities about the suitability of the file's contents. > > > > Steve > > > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > > This was an incoming fax. I don't have much control over the other > > > end. Why is that error appearing when trying to receive something? > > > > > > Robert > > > > > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > > wrote: > > > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > > >> Please try pdf, you should be in business now. > > >> Please let us know, > > >> > > >> On Jan 18, 2013 12:35 AM, "Robert Craig" > >> > wrote: > > >> > > >> OK, so I've made a little bit of progress. The distant end is a > > >> Windows Fax server hooked up to an analog line. I created a fax > > >> extension in FS, assigned an inbound route to the extension, and > > >> made sure the extension had an email address. I believe the fax > > >> receive wants to complete, but there is an interesting error in > > >> the log about TIF files. Anyone ever seen this before? > > >> > > >> v=0 > > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> c=IN IP4 72.201.112.131 > > >> t=0 0 > > >> m=image 27714 udptl t38 > > >> a=T38FaxVersion:0 > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxFillBitRemoval > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > > >> signal sofia/external/+14792460547 > > >> @flowroute.com [BREAK] > > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > > >> ip/port confirmed. > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > > >> Negotiation Result > > >> ======================================================= > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > > >> status off > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > > >> country: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > > >> vendor: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > > >> model: > > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > > >> processing not successful - result (41) TIFF/F file cannot be > > >> opened.* > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > > >> station id: ATW-4792460547 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > > >> station id: SpanDSP Fax Ident > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > > >> transferred: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > > >> fax pages: 0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > > >> resolution: 0x0 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > > >> Rate: 9600 > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > > >> status off > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > > >> country: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > > >> vendor: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > > >> model: > > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > > >> sofia/external/+14792460547 @flowroute.com > > >> Restore previous codec PCMU:0. > > >> EXECUTE sofia/external/+14792460547 > > >> @flowroute.com hangup() > > >> > > >> Robert > > >> > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> OK, I am not extremely concerned with getting an ATA working > > >> right now. If I could just get an inbound extension-to-email > > >> working it would be great. I'll start from scratch and then let > > >> you know what happens. Just to make sure, I am going to create an > > >> extension, assign an email address to it, make sure there is an > > >> inbound route to that extension, and then test it. > > >> > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > > >> From: gassaad at emassembly.com > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> The reason for outbound extension is: > > >> When you send a fax from ATA to the outside world, it will dial > > >> first to the outbound extension and then after successful receive > > >> it will dial out to the intended number. > > >> > > >> As for the error you have it may be related to your ATA. I am not > > >> sure please see why do you get the following message. > > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > > >> > > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > > >> > wrote: > > >> > > >> OK, I've already tried creating one extension for inbound and > > >> set an email address. Of course, it failed and the errors are > > >> at the bottom of this email. What is the purpose of the > > >> outbound extension? > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > > >> From: gassaad at emassembly.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Sorry for the delay to get back to you. > > >> 1- App --> fax server > > >> 2- Create two extensions (one for inbound) another for > > >> (outbound) From within either extension you can send faxes > > >> "virtual fax" > > >> 3- point your create your outbound and inbound dialplans as > > >> normal but make it to point to those extensions. > > >> > > >> Good luck, > > >> > > >> George > > >> > > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > > >> > > > >> wrote: > > >> > > >> What did you do on the config side to get it to work? > > >> I've tried both an Ata and a fax to email extension and > > >> they both fail in both directions. > > >> > > >> Robert > > >> > > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > > >> > > > >> wrote: > > >> > > >> I use Fusion successfully to send / receive fax. > > >> The way it works: > > >> ATA --->FaxServer. > > >> After successful receipt, Fax Server---> DID where > > >> you have dialled originally. > > >> > > >> Cheers, > > >> > > >> George > > >> > > >> > > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > > >> > >> > wrote: > > >> > > >> Did anyone see the issue or what I may have > > >> missed in the config? As mentioned earlier, I > > >> thought it would after a fresh install. Any help > > >> is appreciated. > > >> > > >> Robert > > >> > > >> > > >> > ------------------------------------------------------------------------ > > >> From: craig.robert at hotmail.com > > >> > > >> To: freeswitch-users at lists.freeswitch.org > > >> > > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > > >> > > >> Subject: Re: [Freeswitch-users] T38 Configuration > > >> > > >> Thanks for the quick response. Below is the > > >> capture from the log viewer. In this test, I am > > >> using a remote fax machine (line from Cox with > > >> physical machine) faxing into the "email server" > > >> where I created an extension for fax-to-email. > > >> Calling number is 1479-246-0547. > > >> > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:99 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> mod_sofia.c:396 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:569 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:559 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State Change > > >> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:556 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate Change HANGUP > > >> -> DOWN > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1510 Close Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_DESTROY] > > >> 2013-01-10 13:39:53.162506 [NOTICE] > > >> switch_core_session.c:1506 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Ended > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1488 Session 26 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Locked, Waiting on > > >> external entities > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:440 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change CS_REPORTING > > >> -> CS_DESTROY > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State > > >> REPORTING going to sleep > > >> 2013-01-10 13:39:53.162506 [DEBUG] > > >> switch_core_state_machine.c:92 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> REPORTING, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:749 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_REPORTING > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:446 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_HANGUP -> CS_REPORTING > > >> > > >> 2013-01-10 13:39:53.066135 [DEBUG] > > >> switch_core_state_machine.c:599 Hangup Command > > >> with no Session system(/usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 > > >> name=+14792460547-2013-01-10-13-39-26 > > >> messages='result: Disconnected after permitted > > >> retries sender:ATW-4792460547 pages:0' > > >> domain=192.168.4.15 caller_id_name='' > > >> caller_id_number=): > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:48 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard > > >> HANGUP, cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:552 Sending BYE to > > >> sofia/external/+14792460547 at flowroute.com > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_sofia.c:503 Channel > > >> sofia/external/+14792460547 at flowroute.com > > >> hanging up, > > >> cause: NORMAL_CLEARING > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:667 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_HANGUP > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:2687 > > >> sofia/external/+14792460547 at flowroute.com > > >> skip receive > > >> message [APPLICATION_EXEC_COMPLETE] (channel is > > >> hungup already) > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:3017 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [KILL] > > >> 2013-01-10 13:39:52.216151 [NOTICE] > > >> mod_dptools.c:1150 Hangup > > >> sofia/external/+14792460547 at flowroute.com > > >> [CS_EXECUTE] > > >> [NORMAL_CLEARING] > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_channel.c:2994 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change ACTIVE -> HANGUP > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> hangup() > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> switch_core_codec.c:244 > > >> sofia/external/+14792460547 at flowroute.com > > >> Restore > > >> previous codec PCMU:0. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:518 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:516 remote model: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:515 remote vendor: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:514 remote country: > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:513 ECM status off > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:509 Total fax pages: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:507 Pages transferred: 0 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > > >> Fax Ident > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:505 Remote station id: > > >> ATW-4792460547 > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:500 Fax processing not > > >> successful - result (48) Disconnected after > > >> permitted retries. > > >> 2013-01-10 13:39:52.216151 [DEBUG] > > >> mod_spandsp_fax.c:487 > > >> > ============================================================================== > > >> > > >> 2013-01-10 13:39:38.766190 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> mod_sofia.c:1487 Remote address:port > > >> [66.42.121.5:31632 ] > > >> has not changed. > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:29.576127 [ERR] > > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > > >> must be between 20 and 10000 > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:29.576127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.566140 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxRateManagement:transferredTCF > > >> m=image 31632 udptl t38 > > >> t=0 0 > > >> c=IN IP4 66.42.121.5 > > >> s=- > > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > > >> v=0 > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > > >> Remote SDP: > > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completing][200] > > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > > >> sofia/external/+14792460547 at flowroute.com > > >> Update Callee > > >> ID to "+14792460547" <+14792460547> > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.556130 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [calling][0] > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:2647 Local SDP: > > >> > > >> a=T38FaxUdpEC:t38UDPRedundancy > > >> a=T38FaxMaxDatagram:400 > > >> a=T38FaxMaxBuffer:2000 > > >> a=T38FaxRateManagement:transferredTCF > > >> a=T38FaxFillBitRemoval > > >> a=T38MaxBitRate:9600 > > >> a=T38FaxVersion:0 > > >> m=image 27588 udptl t38 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:29.256196 [DEBUG] > > >> sofia_glue.c:173 > > >> sofia/external/+14792460547 at flowroute.com > > >> image media sdp: > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1379 Raw write codec activation > > >> Success L16 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_core_codec.c:219 > > >> sofia/external/+14792460547 at flowroute.com > > >> Push codec L16:70 > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> mod_spandsp_fax.c:1363 Raw read codec activation > > >> Success L16 20000 > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > > >> > > >> 2013-01-10 13:39:27.996269 [DEBUG] > > >> switch_ivr_play_say.c:1682 done playing file > > >> silence_stream://2000 > > >> 2013-01-10 13:39:26.256197 [DEBUG] > > >> switch_rtp.c:3624 Correct ip/port confirmed. > > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [ready][200] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.096329 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> switch_ivr_play_say.c:1309 Codec Activated > > >> L16 at 8000hz 1 channels 20ms > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> playback(silence_stream://2000) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > > >> 2013-01-10 13:39:26.016177 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38_request]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38_request=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [fax_enable_t38]=[true] > > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > > >> Channel sofia/external/+14792460547 at flowroute.com > > >> entering state > > >> [completed][200] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(fax_enable_t38=true) > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:975 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:26.006127 [NOTICE] > > >> mod_dptools.c:1176 Channel > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] has been > > >> answered > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_channel.c:3395 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Callstate > > >> Change RINGING -> ACTIVE > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> > > >> a=sendrecv > > >> a=ptime:20 > > >> a=silenceSupp:off - - - - > > >> a=fmtp:101 0-16 > > >> a=rtpmap:101 telephone-event/8000 > > >> a=rtpmap:0 PCMU/8000 > > >> m=audio 27588 RTP/AVP 0 101 > > >> t=0 0 > > >> c=IN IP4 72.201.112.131 > > >> s=FreeSWITCH > > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > > >> 72.201.112.131 > > >> v=0 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> mod_sofia.c:856 Local SDP > > >> sofia/external/+14792460547 at flowroute.com > > >> : > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3645 > > >> sofia/external/+14792460547 at flowroute.com > > >> Set rtp dtmf > > >> delay to 40 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > > >> to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > > >> per 20ms > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:3348 AUDIO RTP > > >> [sofia/external/+14792460547 at flowroute.com > > >> ] 192.168.4.15 > > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > > >> 2013-01-10 13:39:26.006127 [DEBUG] > > >> sofia_glue.c:1046 STUN Success > > >> [72.201.112.131]:[27588] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> answer() > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [api_hangup_hook]=[system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=${last_fax} messages='result: > > >> ${fax_result_text} > > >> sender:${fax_remote_station_id} > > >> pages:${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='${caller_id_name}' > > >> caller_id_number=${caller_id_number}] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\${last_fax} > > >> messages='result: \${fax_result_text} > > >> sender:\${fax_remote_station_id} > > >> pages:\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\${caller_id_name}' > > >> caller_id_number=\${caller_id_number}) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> > hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > > >> > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> hash(insert/192.168.4.15-spymap/+14792460547 > /d276f012-5b65-11e2-beae-8564ff827a25) > > >> > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [call_direction]=[local] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(call_direction=local) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action hangup() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> playback(silence_stream://2000) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38_request=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(fax_enable_t38=true) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action answer() > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(api_hangup_hook=system /usr/bin/php > > >> /var/www/html/fusionpbx/secure/fax_to_email.php > > >> email=robert.l.craig at craignetwork.com > > >> > > >> extension=3150 name=\\\${last_fax} > > >> messages='result: \\\${fax_result_text} > > >> sender:\\\${fax_remote_station_id} > > >> pages:\\\${fax_document_total_pages}' > > >> domain=192.168.4.15 > > >> caller_id_name='\\\${caller_id_name}' > > >> caller_id_number=\\\${caller_id_number}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test Fax] destination_number(3150) =~ > > >> /^3150$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Test Fax] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain_user] destination_number(3150) =~ > > >> /^\*97$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain_user] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [vmain] destination_number(3150) =~ > > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->vmain] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [send_to_voicemail] > > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->send_to_voicemail] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [extension-intercom] > > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->extension-intercom] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [intercept-ext] destination_number(3150) > > >> =~ /^\*\*(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->intercept-ext] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_return] destination_number(3150) =~ > > >> /^\*69$|^lcr$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_return] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call_privacy] destination_number(3150) =~ > > >> /^\*67(\d+)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call_privacy] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > > >> > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Absolute > > >> Condition [global] > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > > >> break=never > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->global] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [redial] destination_number(3150) =~ > > >> /^(redial|\*870)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->redial] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [group-intercept] destination_number(3150) > > >> =~ /^\*8$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->group-intercept] continue=false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> ANTI-Action > > >> set(call_direction=local) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [call-direction] ${call_direction}() =~ > > >> /^(inbound|outbound|local)$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->call-direction] continue=true > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (FAIL) [Flowroute.11d] destination_number(3150) > > >> =~ /^\+?(\d{11})$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > >> [default->Flowroute.11d] continue=false > > >> 2013-01-10 13:39:25.816280 [INFO] > > >> mod_dialplan_xml.c:498 Processing +14792460547 > > >> <+14792460547>->3150 in context default > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:117 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:149 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_ROUTING > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [NOTICE] > > >> switch_ivr.c:1781 Transfer > > >> sofia/external/+14792460547 at flowroute.com > > >> to > > >> XML[3150 at default] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:830 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_ivr.c:1774 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_EXECUTE -> CS_ROUTING > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> transfer(3150 > > >> XML default) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_channel.c:1135 EXPORT (export_vars) > > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_dptools.c:1344 > > >> sofia/external/+14792460547 at flowroute.com > > >> SET > > >> [outside_call]=[true] > > >> EXECUTE sofia/external/+14792460547 at flowroute.com > > >> > > >> set(outside_call=true) > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:209 > > >> sofia/external/+14792460547 at flowroute.com > > >> Standard EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> mod_sofia.c:242 > > >> sofia/external/+14792460547 at flowroute.com > > >> SOFIA EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:477 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:415 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) Running State > > >> Change CS_EXECUTE > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:470 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State ROUTING > > >> going to sleep > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_session.c:1283 Send signal > > >> sofia/external/+14792460547 at flowroute.com > > >> [BREAK] > > >> 2013-01-10 13:39:25.816280 [DEBUG] > > >> switch_core_state_machine.c:167 > > >> (sofia/external/+14792460547 at flowroute.com > > >> ) State Change > > >> CS_ROUTING -> CS_EXECUTE > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Action > > >> transfer(3150 XML default) > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] > > >> destination_number(15203039810) =~ > > >> /^15203039810$/ break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> Regex > > >> (PASS) [Test_Number] context(public) =~ /public/ > > > >> break=on-false > > >> Dialplan: > > >> sofia/external/+14792460547 at flowroute.com > > >> parsing > > ... > > [Message clipped] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/70761173/attachment-0001.html From craig.robert at hotmail.com Tue Jan 22 01:54:50 2013 From: craig.robert at hotmail.com (Robert Craig) Date: Mon, 21 Jan 2013 15:54:50 -0700 Subject: [Freeswitch-users] T38 Configuration In-Reply-To: References: , , , , , , , , , , , , , <50F975E5.2040206@coppice.org>, , , , <5F8668D1-AB71-449A-8041-65FAB34C74BC@endigotech.com>, , , , , , Message-ID: Gotcha... Robert Date: Mon, 21 Jan 2013 19:49:40 -0300 From: jmesquita at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Just a correction.. Faxing is NOT a simple thing. Specially when it is dealing with VoIP systems...Jo?o Mesquita FreeSWITCH? Solutions On Mon, Jan 21, 2013 at 6:10 PM, Robert Craig wrote: OK, so I'll change the permissions to include read/write/execute. FS is running as root because this is a box at the house I am playing with. I installed FS using one of the script methods out there and I was logged in as root at the time when I ran everything. Regardless, I'm not a fan of the script install because as you can see, a simple feature such as faxing has created headaches. I didn't think it would be this difficult to turn on fax to email after a default install out of the box. My Asterisk box has given me nothing but headaches when it comes to fax signals so I figured I would give FS a try in that aspect. Once I change the permissions, I'll reboot the thing just to make sure and then run another test. If this works, I am going to completely rebuild this thing and hopefully the folder creation problem goes away as well. When I get to that point, what source/guide do you guys recommend for a manual install? Robert From: bdfoster at endigotech.com Date: Mon, 21 Jan 2013 14:27:37 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Why are you running Freeswitch as root? Sent from my iPhone On Jan 21, 2013, at 12:59 AM, Robert Craig wrote: As far as I know, FS is running as root. Below are the permissions. <13>:ls -ld storage/fax/3150/inbox drw-rw-rw- 2 root root 4096 Jan 18 10:28 storage/fax/3150/inbox I've gone from the 'freeswitch' folder all the way into the inbox folder verifying the same permissions down the line. Should it include 'execute' permissions as well? Whats even more interesting is that when I create a new fax extension, it doesn't create the folder, such as 3150. I have to manually create it. Robert Date: Sun, 20 Jan 2013 17:07:50 -0300 From: jmesquita at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Robert, you clearly still have a permissions problem on this specific folder (according to your dialplan): /usr/local/freeswitch/storage/fax/3150/inbox/ What is the user FreeSWITCH is running under and what are the current permissions on the folder? Jo?o Mesquita FreeSWITCH? Solutions On Sun, Jan 20, 2013 at 3:36 PM, Robert Craig wrote: George, I'm having a hard time understanding your dialplan. I'm not a FS guru so my apologies. It looks like your dialplan is meant for an ATA. I am just trying to get the email portion configured first; specifically inbound. Once I resolve the inbound issues, I'll work on outbound and then move to ATA's. Robert Date: Sun, 20 Jan 2013 09:52:11 -0500 From: gassaad at emassembly.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Here is mine: Inbound:http://pastebin.freeswitch.org/20471Outbound:http://pastebin.freeswitch.org/20470 Thanks,George On Sat, Jan 19, 2013 at 2:54 PM, Robert Craig wrote: OK, I copied everything from the fax extension dial plan into the below pastebin. http://pastebin.freeswitch.org/20466 Robert From: bdfoster at endigotech.com Date: Fri, 18 Jan 2013 12:59:38 -0500 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration Submit the dialplan you are using to receive the fax. You can use the FreeSWITCH pastebin, http://pastebin.freeswitch.org Sent from my iPhone On Jan 18, 2013, at 12:43 PM, Robert Craig wrote: I believe I'm just using the built in SpanDSP for faxing. I've looked around for 'rxfax' and can't find anything. I did notice that the directory where the faxes were supposed to be stored for the extension I create didn't even exist. I created them, applied the permissions, but it seems like I am missing something because the same error is still there. Robert Date: Fri, 18 Jan 2013 09:13:36 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] T38 Configuration If you are using the rxfax app then you specify the file name and location: Make sure that the target directory has read/write permissions. You can use something like "chmod 666 /my/directory" to make it read/write. -MC On Fri, Jan 18, 2013 at 8:26 AM, Robert Craig wrote: Where does the file get downloaded to? What should the permissions be? Robert > Date: Sat, 19 Jan 2013 00:18:45 +0800 > From: steveu at coppice.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] T38 Configuration > > If the TIFF file cannot be opened on receive you probably have a file > permissions issue (e.g. with the directory in which the file is to be > created). If the TIFF file cannot be opened on transmit there are > various possibilities about the suitability of the file's contents. > > Steve > > On 01/18/2013 09:37 PM, Robert Craig wrote: > > This was an incoming fax. I don't have much control over the other > > end. Why is that error appearing when trying to receive something? > > > > Robert > > > > > > > > On Jan 18, 2013, at 5:58 AM, "George Assaad" > > wrote: > > > >> Although the gui mentioned to attach a tiff, I finds it to not work. > >> Please try pdf, you should be in business now. > >> Please let us know, > >> > >> On Jan 18, 2013 12:35 AM, "Robert Craig" >> > wrote: > >> > >> OK, so I've made a little bit of progress. The distant end is a > >> Windows Fax server hooked up to an analog line. I created a fax > >> extension in FS, assigned an inbound route to the extension, and > >> made sure the extension had an email address. I believe the fax > >> receive wants to complete, but there is an interesting error in > >> the log about TIF files. Anyone ever seen this before? > >> > >> v=0 > >> o=FreeSWITCH 1358459217 1358459221 IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> c=IN IP4 72.201.112.131 > >> t=0 0 > >> m=image 27714 udptl t38 > >> a=T38FaxVersion:0 > >> a=T38MaxBitRate:9600 > >> a=T38FaxFillBitRemoval > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxUdpEC:t38UDPRedundancy > >> > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:28:55.315352 [DEBUG] switch_core_session.c:975 Send > >> signal sofia/external/+14792460547 > >> @flowroute.com [BREAK] > >> 2013-01-17 22:29:03.655330 [DEBUG] switch_rtp.c:3624 Correct > >> ip/port confirmed. > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:327 === > >> Negotiation Result > >> ======================================================= > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:328 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:329 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:330 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:332 ECM > >> status off > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:333 remote > >> country: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:334 remote > >> vendor: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:335 remote > >> model: > >> 2013-01-17 22:29:05.695399 [DEBUG] mod_spandsp_fax.c:337 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> *2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:500 Fax > >> processing not successful - result (41) TIFF/F file cannot be > >> opened.* > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:505 Remote > >> station id: ATW-4792460547 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:506 Local > >> station id: SpanDSP Fax Ident > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:507 Pages > >> transferred: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:509 Total > >> fax pages: 0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:510 Image > >> resolution: 0x0 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:511 Transfer > >> Rate: 9600 > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:513 ECM > >> status off > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:514 remote > >> country: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:515 remote > >> vendor: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:516 remote > >> model: > >> 2013-01-17 22:29:08.195304 [DEBUG] mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-17 22:29:08.195304 [DEBUG] switch_core_codec.c:244 > >> sofia/external/+14792460547 @flowroute.com > >> Restore previous codec PCMU:0. > >> EXECUTE sofia/external/+14792460547 > >> @flowroute.com hangup() > >> > >> Robert > >> > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Wed, 16 Jan 2013 10:40:31 -0700 > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> OK, I am not extremely concerned with getting an ATA working > >> right now. If I could just get an inbound extension-to-email > >> working it would be great. I'll start from scratch and then let > >> you know what happens. Just to make sure, I am going to create an > >> extension, assign an email address to it, make sure there is an > >> inbound route to that extension, and then test it. > >> > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 12:11:34 -0500 > >> From: gassaad at emassembly.com > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> The reason for outbound extension is: > >> When you send a fax from ATA to the outside world, it will dial > >> first to the outbound extension and then after successful receive > >> it will dial out to the intended number. > >> > >> As for the error you have it may be related to your ATA. I am not > >> sure please see why do you get the following message. > >> "Invalid Jitterbuffer spec [0] must be between 20 and 10000 " > >> > >> On Wed, Jan 16, 2013 at 11:54 AM, Robert Craig > >> > wrote: > >> > >> OK, I've already tried creating one extension for inbound and > >> set an email address. Of course, it failed and the errors are > >> at the bottom of this email. What is the purpose of the > >> outbound extension? > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> Date: Wed, 16 Jan 2013 09:57:51 -0500 > >> From: gassaad at emassembly.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Sorry for the delay to get back to you. > >> 1- App --> fax server > >> 2- Create two extensions (one for inbound) another for > >> (outbound) From within either extension you can send faxes > >> "virtual fax" > >> 3- point your create your outbound and inbound dialplans as > >> normal but make it to point to those extensions. > >> > >> Good luck, > >> > >> George > >> > >> On Sat, Jan 12, 2013 at 4:05 PM, Robert Craig > >> > > >> wrote: > >> > >> What did you do on the config side to get it to work? > >> I've tried both an Ata and a fax to email extension and > >> they both fail in both directions. > >> > >> Robert > >> > >> On Jan 12, 2013, at 2:04 PM, "George Assaad" > >> > > >> wrote: > >> > >> I use Fusion successfully to send / receive fax. > >> The way it works: > >> ATA --->FaxServer. > >> After successful receipt, Fax Server---> DID where > >> you have dialled originally. > >> > >> Cheers, > >> > >> George > >> > >> > >> On Fri, Jan 11, 2013 at 4:31 PM, Robert Craig > >> >> > wrote: > >> > >> Did anyone see the issue or what I may have > >> missed in the config? As mentioned earlier, I > >> thought it would after a fresh install. Any help > >> is appreciated. > >> > >> Robert > >> > >> > >> ------------------------------------------------------------------------ > >> From: craig.robert at hotmail.com > >> > >> To: freeswitch-users at lists.freeswitch.org > >> > >> Date: Thu, 10 Jan 2013 13:43:09 -0700 > >> > >> Subject: Re: [Freeswitch-users] T38 Configuration > >> > >> Thanks for the quick response. Below is the > >> capture from the log viewer. In this test, I am > >> using a remote fax machine (line from Cox with > >> physical machine) faxing into the "email server" > >> where I created an extension for fax-to-email. > >> Calling number is 1479-246-0547. > >> > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:99 > >> sofia/external/+14792460547 at flowroute.com > >> Standard DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> mod_sofia.c:396 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:569 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:559 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State Change > >> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:556 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate Change HANGUP > >> -> DOWN > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1510 Close Channel > >> sofia/external/+14792460547 at flowroute.com > >> [CS_DESTROY] > >> 2013-01-10 13:39:53.162506 [NOTICE] > >> switch_core_session.c:1506 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Ended > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1488 Session 26 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Locked, Waiting on > >> external entities > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:440 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change CS_REPORTING > >> -> CS_DESTROY > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State > >> REPORTING going to sleep > >> 2013-01-10 13:39:53.162506 [DEBUG] > >> switch_core_state_machine.c:92 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> REPORTING, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:749 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_REPORTING > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:446 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_HANGUP -> CS_REPORTING > >> > >> 2013-01-10 13:39:53.066135 [DEBUG] > >> switch_core_state_machine.c:599 Hangup Command > >> with no Session system(/usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 > >> name=+14792460547-2013-01-10-13-39-26 > >> messages='result: Disconnected after permitted > >> retries sender:ATW-4792460547 pages:0' > >> domain=192.168.4.15 caller_id_name='' > >> caller_id_number=): > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:48 > >> sofia/external/+14792460547 at flowroute.com > >> Standard > >> HANGUP, cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:552 Sending BYE to > >> sofia/external/+14792460547 at flowroute.com > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_sofia.c:503 Channel > >> sofia/external/+14792460547 at flowroute.com > >> hanging up, > >> cause: NORMAL_CLEARING > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:667 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_HANGUP > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:2687 > >> sofia/external/+14792460547 at flowroute.com > >> skip receive > >> message [APPLICATION_EXEC_COMPLETE] (channel is > >> hungup already) > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:3017 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [KILL] > >> 2013-01-10 13:39:52.216151 [NOTICE] > >> mod_dptools.c:1150 Hangup > >> sofia/external/+14792460547 at flowroute.com > >> [CS_EXECUTE] > >> [NORMAL_CLEARING] > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_channel.c:2994 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change ACTIVE -> HANGUP > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> hangup() > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> switch_core_codec.c:244 > >> sofia/external/+14792460547 at flowroute.com > >> Restore > >> previous codec PCMU:0. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:518 > >> ============================================================================== > >> > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:516 remote model: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:515 remote vendor: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:514 remote country: > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:513 ECM status off > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:511 Transfer Rate: 14400 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:510 Image resolution: 0x0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:509 Total fax pages: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:507 Pages transferred: 0 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:506 Local station id: SpanDSP > >> Fax Ident > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:505 Remote station id: > >> ATW-4792460547 > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:500 Fax processing not > >> successful - result (48) Disconnected after > >> permitted retries. > >> 2013-01-10 13:39:52.216151 [DEBUG] > >> mod_spandsp_fax.c:487 > >> ============================================================================== > >> > >> 2013-01-10 13:39:38.766190 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822782 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822781 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> mod_sofia.c:1487 Remote address:port > >> [66.42.121.5:31632 ] > >> has not changed. > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:29.576127 [ERR] > >> sofia_glue.c:3549 Invalid Jitterbuffer spec [0] > >> must be between 20 and 10000 > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:29.576127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.566140 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxRateManagement:transferredTCF > >> m=image 31632 udptl t38 > >> t=0 0 > >> c=IN IP4 66.42.121.5 > >> s=- > >> o=- 275426305 1357849876 IN IP4 66.42.121.5 > >> v=0 > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5614 > >> Remote SDP: > >> 2013-01-10 13:39:29.566140 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completing][200] > >> 2013-01-10 13:39:29.566140 [INFO] sofia.c:931 > >> sofia/external/+14792460547 at flowroute.com > >> Update Callee > >> ID to "+14792460547" <+14792460547> > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.556130 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:29.266195 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [calling][0] > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:2647 Local SDP: > >> > >> a=T38FaxUdpEC:t38UDPRedundancy > >> a=T38FaxMaxDatagram:400 > >> a=T38FaxMaxBuffer:2000 > >> a=T38FaxRateManagement:transferredTCF > >> a=T38FaxFillBitRemoval > >> a=T38MaxBitRate:9600 > >> a=T38FaxVersion:0 > >> m=image 27588 udptl t38 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822780 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:29.256196 [DEBUG] > >> sofia_glue.c:173 > >> sofia/external/+14792460547 at flowroute.com > >> image media sdp: > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1379 Raw write codec activation > >> Success L16 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_core_codec.c:219 > >> sofia/external/+14792460547 at flowroute.com > >> Push codec L16:70 > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> mod_spandsp_fax.c:1363 Raw read codec activation > >> Success L16 20000 > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/+14792460547-2013-01-10-13-39-26.tif) > >> > >> 2013-01-10 13:39:27.996269 [DEBUG] > >> switch_ivr_play_say.c:1682 done playing file > >> silence_stream://2000 > >> 2013-01-10 13:39:26.256197 [DEBUG] > >> switch_rtp.c:3624 Correct ip/port confirmed. > >> 2013-01-10 13:39:26.116239 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [ready][200] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.096329 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> switch_ivr_play_say.c:1309 Codec Activated > >> L16 at 8000hz 1 channels 20ms > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> playback(silence_stream://2000) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [last_fax]=[+14792460547-2013-01-10-13-39-26] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(last_fax=+14792460547-2013-01-10-13-39-26) > >> 2013-01-10 13:39:26.016177 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38_request]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38_request=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [fax_enable_t38]=[true] > >> 2013-01-10 13:39:26.006127 [DEBUG] sofia.c:5603 > >> Channel sofia/external/+14792460547 at flowroute.com > >> entering state > >> [completed][200] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(fax_enable_t38=true) > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:975 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:26.006127 [NOTICE] > >> mod_dptools.c:1176 Channel > >> [sofia/external/+14792460547 at flowroute.com > >> ] has been > >> answered > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_channel.c:3395 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Callstate > >> Change RINGING -> ACTIVE > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> > >> a=sendrecv > >> a=ptime:20 > >> a=silenceSupp:off - - - - > >> a=fmtp:101 0-16 > >> a=rtpmap:101 telephone-event/8000 > >> a=rtpmap:0 PCMU/8000 > >> m=audio 27588 RTP/AVP 0 101 > >> t=0 0 > >> c=IN IP4 72.201.112.131 > >> s=FreeSWITCH > >> o=FreeSWITCH 1357822778 1357822779 IN IP4 > >> 72.201.112.131 > >> v=0 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> mod_sofia.c:856 Local SDP > >> sofia/external/+14792460547 at flowroute.com > >> : > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3645 > >> sofia/external/+14792460547 at flowroute.com > >> Set rtp dtmf > >> delay to 40 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3618 Set 2833 dtmf receive payload > >> to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3612 Set 2833 dtmf send payload to 101 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> switch_rtp.c:1928 Starting timer [soft] 160 bytes > >> per 20ms > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:3348 AUDIO RTP > >> [sofia/external/+14792460547 at flowroute.com > >> ] 192.168.4.15 > >> port 27588 -> 66.42.121.5 port 31632 codec: 0 ms: 20 > >> 2013-01-10 13:39:26.006127 [DEBUG] > >> sofia_glue.c:1046 STUN Success > >> [72.201.112.131]:[27588] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> answer() > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [api_hangup_hook]=[system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=${last_fax} messages='result: > >> ${fax_result_text} > >> sender:${fax_remote_station_id} > >> pages:${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='${caller_id_name}' > >> caller_id_number=${caller_id_number}] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\${last_fax} > >> messages='result: \${fax_result_text} > >> sender:\${fax_remote_station_id} > >> pages:\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\${caller_id_name}' > >> caller_id_number=\${caller_id_number}) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/global/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-last_dial/+14792460547/3150) > >> > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> hash(insert/192.168.4.15-spymap/+14792460547/d276f012-5b65-11e2-beae-8564ff827a25) > >> > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [call_direction]=[local] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(call_direction=local) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action hangup() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> rxfax(/usr/local/freeswitch/storage/fax/3150/inbox/${last_fax}.tif) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> playback(silence_stream://2000) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(last_fax=${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38_request=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(fax_enable_t38=true) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action answer() > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(api_hangup_hook=system /usr/bin/php > >> /var/www/html/fusionpbx/secure/fax_to_email.php > >> email=robert.l.craig at craignetwork.com > >> > >> extension=3150 name=\\\${last_fax} > >> messages='result: \\\${fax_result_text} > >> sender:\\\${fax_remote_station_id} > >> pages:\\\${fax_document_total_pages}' > >> domain=192.168.4.15 > >> caller_id_name='\\\${caller_id_name}' > >> caller_id_number=\\\${caller_id_number}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test Fax] destination_number(3150) =~ > >> /^3150$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Test Fax] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain_user] destination_number(3150) =~ > >> /^\*97$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain_user] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [vmain] destination_number(3150) =~ > >> /^vmain$|^\*4000$|^\*98$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->vmain] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [send_to_voicemail] > >> destination_number(3150) =~ /^\*99(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->send_to_voicemail] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [extension-intercom] > >> destination_number(3150) =~ /^\*8(\d{2,7})$/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->extension-intercom] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [intercept-ext] destination_number(3150) > >> =~ /^\*\*(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->intercept-ext] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_return] destination_number(3150) =~ > >> /^\*69$|^lcr$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_return] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call_privacy] destination_number(3150) =~ > >> /^\*67(\d+)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call_privacy] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/global/${uuid}) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > >> > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Absolute > >> Condition [global] > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [global] ${call_debug}(false) =~ /^true$/ > >> break=never > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->global] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [redial] destination_number(3150) =~ > >> /^(redial|\*870)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->redial] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [group-intercept] destination_number(3150) > >> =~ /^\*8$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->group-intercept] continue=false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> ANTI-Action > >> set(call_direction=local) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [call-direction] ${call_direction}() =~ > >> /^(inbound|outbound|local)$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->call-direction] continue=true > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (FAIL) [Flowroute.11d] destination_number(3150) > >> =~ /^\+?(\d{11})$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing > >> [default->Flowroute.11d] continue=false > >> 2013-01-10 13:39:25.816280 [INFO] > >> mod_dialplan_xml.c:498 Processing +14792460547 > >> <+14792460547>->3150 in context default > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:117 > >> sofia/external/+14792460547 at flowroute.com > >> Standard ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:149 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_ROUTING > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> going to sleep > >> 2013-01-10 13:39:25.816280 [NOTICE] > >> switch_ivr.c:1781 Transfer > >> sofia/external/+14792460547 at flowroute.com > >> to > >> XML[3150 at default] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:830 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_ivr.c:1774 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_EXECUTE -> CS_ROUTING > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> transfer(3150 > >> XML default) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_channel.c:1135 EXPORT (export_vars) > >> [RFC2822_DATE]=[Thu, 10 Jan 2013 13:39:25 -0700] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> export(RFC2822_DATE=Thu, 10 Jan 2013 13:39:25 -0700) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_dptools.c:1344 > >> sofia/external/+14792460547 at flowroute.com > >> SET > >> [outside_call]=[true] > >> EXECUTE sofia/external/+14792460547 at flowroute.com > >> > >> set(outside_call=true) > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:209 > >> sofia/external/+14792460547 at flowroute.com > >> Standard EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> mod_sofia.c:242 > >> sofia/external/+14792460547 at flowroute.com > >> SOFIA EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:477 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:415 > >> (sofia/external/+14792460547 at flowroute.com > >> ) Running State > >> Change CS_EXECUTE > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:470 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State ROUTING > >> going to sleep > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_session.c:1283 Send signal > >> sofia/external/+14792460547 at flowroute.com > >> [BREAK] > >> 2013-01-10 13:39:25.816280 [DEBUG] > >> switch_core_state_machine.c:167 > >> (sofia/external/+14792460547 at flowroute.com > >> ) State Change > >> CS_ROUTING -> CS_EXECUTE > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Action > >> transfer(3150 XML default) > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] > >> destination_number(15203039810) =~ > >> /^15203039810$/ break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> Regex > >> (PASS) [Test_Number] context(public) =~ /public/ > >> break=on-false > >> Dialplan: > >> sofia/external/+14792460547 at flowroute.com > >> parsing ... [Message clipped] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/547f1c46/attachment-0001.html From lists at telefaks.de Tue Jan 22 01:56:09 2013 From: lists at telefaks.de (Peter Steinbach) Date: Mon, 21 Jan 2013 23:56:09 +0100 Subject: [Freeswitch-users] mod_stress In-Reply-To: References: Message-ID: <50FDC789.9000707@telefaks.de> I am also wondering, what this module does in detail. Regarding the source code, I can see that FFT's are made in order to to get some sort of frequency profile. But the code and remarks do not give me a clue how this module actually works. Best regards Peter On 05/26/12 15:05, BF wrote: > Hello, > > I'd like to know what the "stress write" from mod_stress module does. I have hard time finding documentation or an example about mod_stress module. > > Anyone? > > Thank you. > > Eric > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From anthony.minessale at gmail.com Tue Jan 22 02:30:45 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 Jan 2013 17:30:45 -0600 Subject: [Freeswitch-users] mod_stress In-Reply-To: <50FDC789.9000707@telefaks.de> References: <50FDC789.9000707@telefaks.de> Message-ID: That module is incomplete we made it several years ago to test some code that was supposed to pick up on stress in your voice in a media bug so we could fire events and whisper "lie" to the caller as he talked to the guy. On Mon, Jan 21, 2013 at 4:56 PM, Peter Steinbach wrote: > I am also wondering, what this module does in detail. > > Regarding the source code, I can see that FFT's are made in order to to > get some sort of frequency profile. > But the code and remarks do not give me a clue how this module actually > works. > > Best regards Peter > > > > On 05/26/12 15:05, BF wrote: > > Hello, > > > > I'd like to know what the "stress write" from mod_stress module does. I > have hard time finding documentation or an example about mod_stress module. > > > > Anyone? > > > > Thank you. > > > > Eric > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/d1991ce3/attachment.html From msc at freeswitch.org Tue Jan 22 03:49:09 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 16:49:09 -0800 Subject: [Freeswitch-users] Make a call with Perl In-Reply-To: References: <50FD5C8B.9030306@iway.ch> Message-ID: Or the awesome example from the FreeSWITCH Cookbook... ;) -MC On Mon, Jan 21, 2013 at 2:17 PM, Hector Geraldino < Hector.Geraldino at ipsoft.com> wrote: > I'm not a perl expert, but in general I think you need to follow this > pattern: > > 1) Establish the connection to FreeSWITCH, you'll be using the event > socket on inbound mode > 2) Start listening for the events you want to capture (send the 'event > plain all' or 'event plain CHANNEL_ANSWER DTMF CHANNEL_HANGUP ...' command) > 3) Create an uuid (by sending the create_uuid command) so you can identify > which call you'll be listening events for > 4) Add a filter for this uuid, so you'll only receive events for this call > over this connection (filter Unique-ID + uuid) > 5) originate the call: {origination_uuid= + uuid}sofia/internal/NXX at GATEWAY > > Check this simple perl example: > > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Simple_Perl_Example > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matthias > Cramer > Sent: Monday, January 21, 2013 10:20 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Make a call with Perl > > Hi All > > I try to make a Perl Script which calls a certain external Phone number. > It then should wait until someone picks up and play a sound file The user > on the other end then needs to confirm the call with a pin Then another > sound file should be played and the the call should be terminated. > > I tried several ways with ESL and originate. But I allays end up with a > call I have no control over. I can read DTMF but can not hang up the call > or play a second file. > > I think I am going in to a wrong direction. > > Any hints ? > > Regards > > Matthias > > -- > Matthias Cramer / mc322-ripe Senior Network & Security Engineer > iway AG Phone +41 43 500 1111 > Badenerstrasse 569 Fax +41 44 271 3535 > CH-8048 Z?rich http://www.iway.ch/ > GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/456dc6a1/attachment.html From msc at freeswitch.org Tue Jan 22 04:06:36 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Jan 2013 17:06:36 -0800 Subject: [Freeswitch-users] Portaudio troubles In-Reply-To: <50FDAB3A.5050602@orresta.no-ip.com> References: <50FDAB3A.5050602@orresta.no-ip.com> Message-ID: A core dump is always worthy of a Jira if you are able to reproduce it on the latest git HEAD. First, though, make sure that endpoint "out-right" exists in conf/autoload_configs/portaudio.conf.xml and points to a real device on the system. Be sure to use "pa devlist" to see what's really there. -MC On Mon, Jan 21, 2013 at 12:55 PM, Jakob wrote: > Hi, > > I'm trying to get working a line into en external PA system > > Whenever i run this extension > > break="on-true"> > data="portaudio/endpoint/out-right"/> > > > Freeswitch coredumps and has to be restarted. > > The log for the last call in the console before it goes dead follows > > EXECUTE sofia/external/1050 at rshaninge.mress.se > bridge(portaudio/endpoint/out-right) > 2013-01-21 21:46:45.057266 [DEBUG] switch_ivr_originate.c:2022 Parsing > global variables > 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:2362 Created shared > audio stream alpha: 48000 channels 2 > 2013-01-21 21:46:45.576716 [NOTICE] switch_channel.c:968 New Channel > portaudio/endpoint-out-right [d9f50f6b-ea97-47a9-b721-659c1a40af14] > 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:1272 > (portaudio/endpoint-out-right) State Change CS_NEW -> CS_INIT > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:1283 Send > signal portaudio/endpoint-out-right [BREAK] > 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:804 > portaudio/endpoint-out-right CHANNEL KILL > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_state_machine.c:415 > (portaudio/endpoint-out-right) Running State Change CS_INIT > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_state_machine.c:454 > (portaudio/endpoint-out-right) State INIT > 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:286 > (portaudio/endpoint-out-right) State Change CS_INIT -> CS_ROUTING > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:1283 Send > signal portaudio/endpoint-out-right [BREAK] > 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:804 > portaudio/endpoint-out-right CHANNEL KILL > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_state_machine.c:454 > (portaudio/endpoint-out-right) State INIT going to sleep > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_state_machine.c:415 > (portaudio/endpoint-out-right) Running State Change CS_ROUTING > 2013-01-21 21:46:45.576716 [DEBUG] switch_channel.c:2003 > (portaudio/endpoint-out-right) Callstate Change DOWN -> RINGING > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_state_machine.c:470 > (portaudio/endpoint-out-right) State ROUTING > 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:666 Added call 1 > 2013-01-21 21:46:45.576716 [DEBUG] switch_channel.c:3395 > (portaudio/endpoint-out-right) Callstate Change RINGING -> ACTIVE > 2013-01-21 21:46:45.576716 [DEBUG] switch_channel.c:3407 Send signal > sofia/external/1050 at rshaninge.mress.se [BREAK] > 2013-01-21 21:46:45.576716 [NOTICE] mod_portaudio.c:360 Channel > [portaudio/endpoint-out-right] has been answered > 2013-01-21 21:46:45.576716 [DEBUG] sofia_glue.c:3351 AUDIO RTP > [sofia/external/1050 at rshaninge.mress.se] 81.216.195.38 port 24486 -> > 10.171.149.115 port 21000 codec: 9 ms: 20 > 2013-01-21 21:46:45.576716 [DEBUG] switch_rtp.c:1940 Starting timer > [soft] 160 bytes per 20ms > 2013-01-21 21:46:45.576716 [DEBUG] sofia_glue.c:3615 Set 2833 dtmf send > payload to 101 > 2013-01-21 21:46:45.576716 [DEBUG] sofia_glue.c:3621 Set 2833 dtmf > receive payload to 101 > 2013-01-21 21:46:45.576716 [DEBUG] sofia_glue.c:3648 > sofia/external/1050 at rshaninge.mress.se Set rtp dtmf delay to 40 > 2013-01-21 21:46:45.576716 [DEBUG] mod_sofia.c:856 Local SDP > sofia/external/1050 at rshaninge.mress.se: > v=0 > o=FreeSWITCH 1358776719 1358776720 IN IP4 81.216.195.38 > s=FreeSWITCH > c=IN IP4 81.216.195.38 > t=0 0 > m=audio 24486 RTP/AVP 9 101 > a=rtpmap:9 G722/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:830 Send signal > sofia/external/1050 at rshaninge.mress.se [BREAK] > 2013-01-21 21:46:45.576716 [DEBUG] switch_channel.c:3395 > (sofia/external/1050 at rshaninge.mress.se) Callstate Change RINGING -> > ACTIVE > 2013-01-21 21:46:45.576716 [NOTICE] switch_ivr_originate.c:3378 Channel > [sofia/external/1050 at rshaninge.mress.se] has been answered > 2013-01-21 21:46:45.576716 [DEBUG] switch_ivr_originate.c:3435 Originate > Resulted in Success: [portaudio/endpoint-out-right] > 2013-01-21 21:46:45.576716 [DEBUG] switch_ivr_bridge.c:1268 > (portaudio/endpoint-out-right) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:1283 Send > signal portaudio/endpoint-out-right [BREAK] > 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:804 > portaudio/endpoint-out-right CHANNEL KILL > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:830 Send signal > portaudio/endpoint-out-right [BREAK] > 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:804 > portaudio/endpoint-out-right CHANNEL KILL > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:830 Send signal > sofia/external/1050 at rshaninge.mress.se [BREAK] > 2013-01-21 21:46:45.576716 [DEBUG] switch_ivr_bridge.c:1370 > (portaudio/endpoint-out-right) State Change CS_CONSUME_MEDIA -> > CS_EXCHANGE_MEDIA > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:1283 Send > signal portaudio/endpoint-out-right [BREAK] > 2013-01-21 21:46:45.576716 [DEBUG] mod_portaudio.c:804 > portaudio/endpoint-out-right CHANNEL KILL > 2013-01-21 21:46:45.576716 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/1050 at rshaninge.mress.se [BREAK] > 2013-01-21 21:46:45.576716 [DEBUG] sofia.c:5599 Channel > sofia/external/1050 at rshaninge.mress.se entering state [completed][200] > 2013-01-21 21:46:45.636815 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/1050 at rshaninge.mress.se [BREAK] > 2013-01-21 21:46:45.636815 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/1050 at rshaninge.mress.se [BREAK] > 2013-01-21 21:46:45.636815 [DEBUG] switch_core_session.c:975 Send signal > sofia/external/1050 at rshaninge.mress.se [BREAK] > 2013-01-21 21:46:45.636815 [DEBUG] sofia.c:5599 Channel > sofia/external/1050 at rshaninge.mress.se entering state [ready][200] > 2013-01-21 21:46:46.576712 [DEBUG] mod_portaudio.c:429 > portaudio/endpoint-out-right CHANNEL ROUTING > 2013-01-21 21:46:46.576712 [DEBUG] switch_core_state_machine.c:470 > (portaudio/endpoint-out-right) State ROUTING going to sleep > 2013-01-21 21:46:46.576712 [DEBUG] switch_core_state_machine.c:415 > (portaudio/endpoint-out-right) Running State Change CS_EXCHANGE_MEDIA > 2013-01-21 21:46:46.576712 [DEBUG] switch_core_state_machine.c:480 > (portaudio/endpoint-out-right) State EXCHANGE_MEDIA > 2013-01-21 21:46:46.576712 [DEBUG] mod_portaudio.c:817 CHANNEL LOOPBACK > 2013-01-21 21:46:46.576712 [DEBUG] switch_core_session.c:892 Send signal > portaudio/endpoint-out-right [BREAK] > 2013-01-21 21:46:46.576712 [DEBUG] mod_portaudio.c:804 > portaudio/endpoint-out-right CHANNEL KILL > 2013-01-21 21:46:46.576712 [DEBUG] switch_core_session.c:892 Send signal > sofia/external/1050 at rshaninge.mress.se [BREAK] > > > Am I missing something in the log that helps in tracking this down. > > Best Regards > Jakob Sundberg > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/3d084b2a/attachment-0001.html From schoch+freeswitch.org at xwin32.com Tue Jan 22 04:17:28 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 21 Jan 2013 17:17:28 -0800 Subject: [Freeswitch-users] MWI doesn't work Message-ID: I just got voicemail working for the first time. (It was broken because of permissions on the /storage directory.) Now the problem is the message waiting indicator. From another extension, I called by extension and waited to be transferred to voicemail. I left myself a message, which was indeed stored as a new message. However, the MWI on my phone did not light. I'm using a Polycom SoundPoint IP 320. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130121/2852f55b/attachment.html From fs-list at communicatefreely.net Tue Jan 22 07:40:55 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 21 Jan 2013 23:40:55 -0500 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: Message-ID: <50FE1857.7020804@communicatefreely.net> Hmm. Good idea. I know that you can do this with Aastra phones, and possibly others. With the Aastra's, there is an XML API that you can use to push notifications, apps, etc. You can push XML to the phone on an HTTP interface (there is an ACL to tell it where to allow it from), and you can initiate phone calls, as well as hang up, transfer, etc. There is also an action URI, so you can have the phone report what is coming in on caller ID, when it is ringing, hung up, etc. I don't know if you can get lamp status from this or not, but perhaps you could have a system where a pattern of calls is generated, and after many repetitions, the system calls or a human to check and see that the lights match an expected state. Just an idea anyway. -Tim Michael Collins wrote: > This test suite does not presently exist. I would love to see some work > done in this area. > > One challenge is that there are numerous kinds of devices that can > interop with FS so the test suite will need to have some way to handle > the various phone makes and models. Also, I don't know which hard > phones, if any, support automated testing. I would *love* to see some > sort of web/network API on various phones where you could poll the phone > to see the state of each SLA/BLF key as well as what's in the display. > That would make automated testing much cleaner. As it stands now, > testing BLFs is a tedious process of watching phones and waiting to see > if the lights do what they should be doing. > > If anyone has practical suggestions (nothing pie-in-the-sky, if you get > my meaning) then by all means share with the group. > > -MC > > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine > > wrote: > > Hello, > > We are running version 1.0.6; we did not upgrade for a long time > due to some problems (mainly stuck BLFs) with the newer versions > which have been corrected a few months ago. > > A few weeks ago I pulled the latest GIT and did functional testings > on our development system; all was OK. > > Yesterday I moved our production system to that version. After a few > hours the problems started: stuck BLFs, hangups, LUA scripts > sometimes do not run. I had to revert back to the old version. > > My question: Is there some simple test suite which I can test > various scnarios on my development system? like, "call XX, while > this call is active call YY which is monitored by XX and hangup", > etc. It is hard to cover all cases manually, so I would like > something automated. > > Thanks, __Yehavi: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vbvbrj at gmail.com Tue Jan 22 10:36:42 2013 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 22 Jan 2013 09:36:42 +0200 Subject: [Freeswitch-users] Reload mod_lua In-Reply-To: References: <50FD7E4F.6080607@gmail.com> Message-ID: <50FE418A.6060207@gmail.com> On 21.01.2013 20:52, Michael Collins wrote: > Lua is not "unloadable" and therefore you cannot do this the way you are > trying to. You'll need to restart FreeSWITCH. > -MC Oh, it's not so good. Update lua scripts that a started within dialplan execution is updating automatically, but scripts which start with FS automatically is not convenient to update only restarting FS. May be there is some method to update lua scripts without restarting FS? -- Mimiko desu. From vbvbrj at gmail.com Tue Jan 22 10:38:09 2013 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 22 Jan 2013 09:38:09 +0200 Subject: [Freeswitch-users] Echo present. In-Reply-To: References: <50FC354E.80206@gmail.com> <50FCE279.2070707@gmail.com> <50FD8ABB.4010809@gmail.com> Message-ID: <50FE41E1.10809@gmail.com> On 21.01.2013 20:44, Michael Collins wrote: > Echo should be removed at the endpoints, not in the middle of the > stream. In most cases, if you hear echo then the other end's echo > canceller is not working, and vice versa. Oh, if this is not my phone's problem, but rather other end's phone/system, then I know who to blame and where to dig. :) Thank you. -- Mimiko desu. From vbvbrj at gmail.com Tue Jan 22 10:40:16 2013 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 22 Jan 2013 09:40:16 +0200 Subject: [Freeswitch-users] Reload mod_lua In-Reply-To: <50FDB776.3060100@quentustech.com> References: <50FD7E4F.6080607@gmail.com> <50FDAFAE.8080608@gmail.com> <50FDB776.3060100@quentustech.com> Message-ID: <50FE4260.3010202@gmail.com> On 21.01.2013 23:47, William King wrote: > Interesting. A mod_lua api that would list the currently running lua > scripts with an id that could be used to kill off individual scripts. > Lua scripts, started with FS on start-up must not be killed, but updated when changing code of script. A killing command for a lua script is ok for scripts that got in loop/. -- Mimiko desu. From slava at tangramltd.com Tue Jan 22 10:45:39 2013 From: slava at tangramltd.com (Viacheslav Dubrovskyi) Date: Tue, 22 Jan 2013 09:45:39 +0200 Subject: [Freeswitch-users] How to enable debug mode for DTMF in FreeTDM? In-Reply-To: <50FD8C6D.3090806@tangramltd.com> References: <50FD8C6D.3090806@tangramltd.com> Message-ID: <50FE43A3.2070104@tangramltd.com> 21.01.2013 20:43, Viacheslav Dubrovskyi ?????: > Hi. > > How to enable debug mode for DTMF in FreeTDM? > > I setup in freetdm.conf > > [general] > debugdtmf_directory=/var/spool/freeswitch/dtmf > > [span zt mySpan] > debugdtmf => yes > > > But there is nothing in path. What else should I do? All ignore me ;( Or is it a secret knowledge available only to the initiated? -- WBR, Viacheslav Dubrovskyi -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3746 bytes Desc: ?????????????????????????????????? ?????????????? S/MIME Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/247e49ef/attachment.bin From slava at tangramltd.com Tue Jan 22 12:16:36 2013 From: slava at tangramltd.com (Viacheslav Dubrovskyi) Date: Tue, 22 Jan 2013 11:16:36 +0200 Subject: [Freeswitch-users] dtmf problem In-Reply-To: <50F38BEE.6090804@synacktics.com> References: <50F38BEE.6090804@synacktics.com> Message-ID: <50FE58F4.2010805@tangramltd.com> 14.01.2013 06:39, Tim Bock ?????: > Hello List, > > I have a freeswitch server with a digium tdm wildcard. Compiled, > enabled, and configured freetdm and dahdi...all seems to be working fine > (can make/receive calls). > > My problem is with dtmf. When calling from an external pstn number to > freeswitch, dtmf seems to be detected properly and consistently. But > when calling from a cell phone or via another voip line, dtmf doesn't > seem to be detected, or detected only occasionally. > > On the same system, if I call into the voip number, calls from any of > the three external options (landline, cell, sip) all seem to work fine > with dtmf. > > So the problem seems to be something I've misconfigured with the tdm > card. Can anyone point me in the right direction? Found this post http://fonality.com/trixbox/forums/trixbox-forums/help/inbound-dtmf-pri-t-analog It seems that explains. But I not sure. -- WBR, Viacheslav Dubrovskyi -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3746 bytes Desc: ?????????????????????????????????? ?????????????? S/MIME Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/cb97b552/attachment-0001.bin From matthias.cramer at iway.ch Tue Jan 22 12:21:15 2013 From: matthias.cramer at iway.ch (Matthias Cramer) Date: Tue, 22 Jan 2013 10:21:15 +0100 Subject: [Freeswitch-users] Make a call with Perl In-Reply-To: References: <50FD5C8B.9030306@iway.ch> Message-ID: <50FE5A0B.1060207@iway.ch> Hi Michael On 21/01/13 20:50, Michael Collins wrote: > If you have the FS Cookbook then check out chapter 4, page 81. If you don't have the FS Cookbook then *SHAME* on you! :D I've ordered the book, but it has not yet arrived... :-( > Tip: you can download the examples from both book at the Packt Publishing website. The only caveat as far as I can tell is that you need to create a login account. You do not need a login... Could download them without one. > Grab the examples for the Cookbook and look at chapter 4. The recipe is called "Launching a call with an inbound event socket connect" and it's written in - you guessed it - Perl. :) I started exactly with that example, I do an originate with &playback(). But I do not have real control over the call. I can not play anything else in the audio stream. And if the playback is finished the call hangs up ... Attached is the script I came up with so far ... Regards Matthias -- Matthias Cramer / mc322-ripe Senior Network & Security Engineer iway AG Phone +41 43 500 1111 Badenerstrasse 569 Fax +41 44 271 3535 CH-8048 Z?rich http://www.iway.ch/ GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250 -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch_call.pl Type: application/x-perl Size: 2582 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/c53eb158/attachment.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 253 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/c53eb158/attachment-0001.bin From lists at telefaks.de Tue Jan 22 14:19:38 2013 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 22 Jan 2013 12:19:38 +0100 Subject: [Freeswitch-users] mod_stress In-Reply-To: References: <50FDC789.9000707@telefaks.de> Message-ID: <50FE75CA.2030007@telefaks.de> Hello Anthony, although it may be incomplete, it may be useful for us. Can you give us some basic explanation, how this works? I can see in the code that there is a stress command with read|write|stop parameter. Is this to be issued via event socket or is it a dialplan application? I suspect that "read" somehow starts the detection and "stop" somehow stops the detection. For "write" I currently have no idea. I would be happy to test this and report my configuration, explanations and examples in the wiki. Best regards Peter On 01/22/13 00:30, Anthony Minessale wrote: > That module is incomplete we made it several years ago to test some > code that was supposed to pick up on stress in your voice in a media > bug so we could fire events and whisper "lie" to the caller as he > talked to the guy. > > > > On Mon, Jan 21, 2013 at 4:56 PM, Peter Steinbach > wrote: > > I am also wondering, what this module does in detail. > > Regarding the source code, I can see that FFT's are made in order > to to > get some sort of frequency profile. > But the code and remarks do not give me a clue how this module > actually > works. > > Best regards Peter > > > > On 05/26/12 15:05, BF wrote: > > Hello, > > > > I'd like to know what the "stress write" from mod_stress module > does. I have hard time finding documentation or an example about > mod_stress module. > > > > Anyone? > > > > Thank you. > > > > Eric > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/4e3b075c/attachment.html From steveayre at gmail.com Tue Jan 22 14:39:18 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 22 Jan 2013 11:39:18 +0000 Subject: [Freeswitch-users] mod_stress In-Reply-To: <50FE75CA.2030007@telefaks.de> References: <50FDC789.9000707@telefaks.de> <50FE75CA.2030007@telefaks.de> Message-ID: Media bugs are used to hook into the call and provide decoded audio to the module. That requires transcoding, naturally. See stress_callback which is called for each audio frame plus some init/cleanup hooks, and read up on the API at http://docs.freeswitch.org/group__mb1.html. I suspect read vs write would correspond to the speaking/listening directions of the channel. Read from caller, write to caller? The names come from the media bug API. The read/write options will add the corresponding media bug, stop will remove them. SWITCH_ADD_APP tells you it's a dialplan application - it's started from dialplan, although you can also trigger dialplan apps on a channel via ESL using uuid_broadcast. The FFT algorithm is used to extract the frequencies present in the sample (the power spectrum). The algorithm then tries to detect stressed based on that. Looks like the algorithm at least in part attempts to look for tremor in the voice. More tremor indicates more stress. Over a threshold it raises an event. Bear in mind that as it's pattern recognition (and little tested at that) you'll get a lot of false positives and false negatives. It's also looking for something in the voice that may or may not be there - some people might be able to speak with an unstressed voice when they're actually quite stressed. -Steve On 22 January 2013 11:19, Peter Steinbach wrote: > Hello Anthony, > > although it may be incomplete, it may be useful for us. Can you give us > some basic explanation, how this works? > I can see in the code that there is a stress command with read|write|stop > parameter. > Is this to be issued via event socket or is it a dialplan application? I > suspect that "read" somehow starts the detection and "stop" somehow stops > the detection. For "write" I currently have no idea. > > I would be happy to test this and report my configuration, explanations > and examples in the wiki. > > Best regards > Peter > > > > On 01/22/13 00:30, Anthony Minessale wrote: > > That module is incomplete we made it several years ago to test some code > that was supposed to pick up on stress in your voice in a media bug so we > could fire events and whisper "lie" to the caller as he talked to the guy. > > > > On Mon, Jan 21, 2013 at 4:56 PM, Peter Steinbach wrote: > >> I am also wondering, what this module does in detail. >> >> Regarding the source code, I can see that FFT's are made in order to to >> get some sort of frequency profile. >> But the code and remarks do not give me a clue how this module actually >> works. >> >> Best regards Peter >> >> >> >> On 05/26/12 15:05, BF wrote: >> > Hello, >> > >> > I'd like to know what the "stress write" from mod_stress module does. I >> have hard time finding documentation or an example about mod_stress module. >> > >> > Anyone? >> > >> > Thank you. >> > >> > Eric >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> -- >> With kind regards >> Peter Steinbach >> >> Telefaks Services GmbH >> mailto:lists (att) telefaks.de >> Internet: www.telefaks.de >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbHmailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/1d652c7d/attachment-0001.html From stargray at bigmir.net Tue Jan 22 14:51:48 2013 From: stargray at bigmir.net (Anton Vojlenko) Date: Tue, 22 Jan 2013 13:51:48 +0200 Subject: [Freeswitch-users] ESL and DEBUG messages Message-ID: Hello, I want to check user status from the Skype client instance, but I can't see Skype API answers from it. How to fix this? Script example: #!/usr/bin/perl use strict; use warnings; require ESL; my $host = "localhost"; my $port = "8021"; my $pass = "somepass"; my $con = new ESL::ESLconnection($host, $port, $pass); $con->api("console loglevel 9"); my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); print $status; $con->disconnect(); Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/39c110bd/attachment.html From jaganthoutam at gmail.com Tue Jan 22 14:55:35 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Tue, 22 Jan 2013 17:25:35 +0530 Subject: [Freeswitch-users] Make a call with Perl In-Reply-To: <50FE5A0B.1060207@iway.ch> References: <50FD5C8B.9030306@iway.ch> <50FE5A0B.1060207@iway.ch> Message-ID: Hi Matthias, You ca use 2 scripts for More control Like 1. will Orginate the call 2. IVR Control #!/usr/bin/perl use DBI; require ESL; my $ph = "1000"; my $dialstr = "jagan.pl"; my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); $con->api("perlrun", "$dialstr $ph"); 1; In jagan.pl You can Control your IVR and what ever you want I hope this will help you Thanks Jagadish On 22 January 2013 14:51, Matthias Cramer wrote: > Hi Michael > > On 21/01/13 20:50, Michael Collins wrote: >> If you have the FS Cookbook then check out chapter 4, page 81. If you don't have the FS Cookbook then *SHAME* on you! :D > > I've ordered the book, but it has not yet arrived... :-( > >> Tip: you can download the examples from both book at the Packt Publishing website. The only caveat as far as I can tell is that you need to create a login account. > > You do not need a login... Could download them without one. > >> Grab the examples for the Cookbook and look at chapter 4. The recipe is called "Launching a call with an inbound event socket connect" and it's written in - you guessed it - Perl. :) > > I started exactly with that example, I do an originate with &playback(). But I do not have real control over the call. I can not play anything else in the audio stream. > > And if the playback is finished the call hangs up ... > > Attached is the script I came up with so far ... > > Regards > > Matthias > > -- > Matthias Cramer / mc322-ripe Senior Network & Security Engineer > iway AG Phone +41 43 500 1111 > Badenerstrasse 569 Fax +41 44 271 3535 > CH-8048 Z?rich http://www.iway.ch/ > GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gvvsubhashkumar at gmail.com Tue Jan 22 15:49:38 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Tue, 22 Jan 2013 18:19:38 +0530 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: Ok. We did the test again,pasted the freeswitch log and uploaded the wireshark of the both success and failed call traces.Please let me know if you have problem in accessing the URLs . http://pastebin.freeswitch.org/20484 http://www.cloudshark.org/captures/6dc99dc2f68c -- Success http://www.cloudshark.org/captures/27b88b2c8ca4--- Failure Thanks, Subhash. On Tue, Jan 22, 2013 at 12:06 AM, Michael Collins wrote: > That may be the case, but that doesn't actually supply the information you > need to find out where the DTMFs are coming from. I would recommend doing > the test that Moy suggested. At least find out where FreeSWITCH thinks this > DTMF is coming from. > > -MC > > > On Sun, Jan 20, 2013 at 6:43 PM, Subhash wrote: > >> But we don't see any problem when both endpoints are SIP. >> >> Thanks, >> Subhash. >> >> >> On Mon, Jan 21, 2013 at 7:12 AM, Moises Silva wrote: >> >>> On Fri, Jan 18, 2013 at 1:27 AM, Subhash wrote: >>> >>>> Yes,we are trying to bridge PSTN and VOIP call. >>>> >>>> We posted the log in pastebin.Please use the below URL to access the >>>> freeswitch log. >>>> >>>> http://pastebin.freeswitch.org/20461 >>>> >>>> >>> >>> >>> 1. 013-01-16 01:02:26.954165 [INFO] ftmod_sangoma_isdn_stack_rcv.c: >>> 144 [s4c1][4:1] Received CONNECT/CONNECT ACK (suId:1 suInstId:2 >>> spInstId:2 ces:0) >>> 2. 2013-01-16 01:02:26.954165 [DEBUG] >>> ftmod_sangoma_isdn_stack_hndl.c:285 [s4c1][4:1] Processing >>> CONNECT/CONNECT ACK (suId:1 suInstId:2 spInstId:2 ces:0) >>> 3. 2013-01-16 01:02:26.974165 [DEBUG] switch_rtp.c:3596 Correct >>> ip/port confirmed. >>> 4. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:1002 [s4c1][4: >>> 1] First packet read stats: Rx queue len: 4, Rx queue size: 10 >>> 5. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:938 [s4c1][4:1 >>> ] First packet write stats: Tx queue len: 1, Tx queue size: 5, Tx >>> idle: 10 >>> 6. 2013-01-16 01:02:27.694163 [DEBUG] switch_rtp.c:3797 RTP RECV >>> DTMF 9:640 >>> 7. 2013-01-16 01:02:27.694163 [DEBUG] switch_ivr_bridge.c:393 Send >>> signal FreeTDM/4:1/2001 [BREAK] >>> 8. 2013-01-16 01:02:27.694163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >>> DTMF [9] >>> >>> >>> The DTMF comes from the RTP side, not FreeTDM (as shown by the log >>> above). You have to verify using tcpdump/wireshark if there is any real >>> DTMF being received. >>> >>> I'd do exactly the same test again, but this time I would also start a >>> wireshark trace. Then please pastebin the trace again and upload the pcap >>> trace to cloudshark (http://www.cloudshark.org/) and paste the link >>> here. >>> >>> Moy >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/e6012685/attachment.html From gmaruzz at gmail.com Tue Jan 22 17:45:06 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 22 Jan 2013 15:45:06 +0100 Subject: [Freeswitch-users] ESL and DEBUG messages In-Reply-To: References: Message-ID: On Tue, Jan 22, 2013 at 12:51 PM, Anton Vojlenko wrote: > Hello, > > I want to check user status from the Skype client instance, but I can't see > Skype API answers from it. How to fix this? Have you tried to give: "fsctl loglevel 9" in addition to "console loglevel 9" ? > Script example: > > #!/usr/bin/perl > use strict; > use warnings; > require ESL; > > my $host = "localhost"; > my $port = "8021"; > my $pass = "somepass"; > my $con = new ESL::ESLconnection($host, $port, $pass); > > $con->api("console loglevel 9"); > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); > print $status; > $con->disconnect(); > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From yehavi.bourvine at gmail.com Tue Jan 22 18:05:55 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 22 Jan 2013 17:05:55 +0200 Subject: [Freeswitch-users] Can a stalled ESL receiver stall FreeSwitch? Message-ID: Hello, Suppose I have a process that connects to Freeswitch via ESL and asks for various high verbosity events. Suppose it does not read the data from the socket - can this stall Freeswitch? Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/ead5fa47/attachment.html From roger.castaldo at gmail.com Tue Jan 22 18:12:06 2013 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Tue, 22 Jan 2013 10:12:06 -0500 Subject: [Freeswitch-users] Can a stalled ESL receiver stall FreeSwitch? In-Reply-To: References: Message-ID: I do not believe it does, if I recall freeswitch will start throwing errors up though stating that the events are not being read and start discarding them because it will only queue up so many. Again not 100% on this but I seem to recall doing some testing a while back that displayed this behavior. On Tue, Jan 22, 2013 at 10:05 AM, Yehavi Bourvine wrote: > Hello, > > Suppose I have a process that connects to Freeswitch via ESL and asks > for various high verbosity events. Suppose it does not read the data from > the socket - can this stall Freeswitch? > > Thanks, __Yehavi: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/cf146aaf/attachment.html From bh at udev.org Tue Jan 22 13:05:48 2013 From: bh at udev.org (Benjamin Henrion) Date: Tue, 22 Jan 2013 11:05:48 +0100 Subject: [Freeswitch-users] VoIP diner at FOSDEM 2013 Message-ID: Hi, I am organizing a diner for all Voice over IP hackers (Asterisk, Freeswitch, Kamailio, OpenSER, Sipvicious :-), etc?) during the Saturday evening of FOSDEM 2013. There is a complete track at the Telephony devroom as well: https://fosdem.org/2013/schedule/track/telephony/ More infos are registration: http://www.zoobab.com/voip-diner-fosdem Best, -- Benjamin Henrion FFII Brussels - +32-484-566109 - +32-2-3500762 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators." From mike at jerris.com Tue Jan 22 18:35:22 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Jan 2013 10:35:22 -0500 Subject: [Freeswitch-users] Echo present. In-Reply-To: <50FE41E1.10809@gmail.com> References: <50FC354E.80206@gmail.com> <50FCE279.2070707@gmail.com> <50FD8ABB.4010809@gmail.com> <50FE41E1.10809@gmail.com> Message-ID: <9343656A-6EA8-4314-96DC-B8F4E9636682@jerris.com> The rule of echo is, if you hear it, its the other guys fault. On Jan 22, 2013, at 2:38 AM, Mimiko wrote: > On 21.01.2013 20:44, Michael Collins wrote: >> Echo should be removed at the endpoints, not in the middle of the >> stream. In most cases, if you hear echo then the other end's echo >> canceller is not working, and vice versa. > > Oh, if this is not my phone's problem, but rather other end's > phone/system, then I know who to blame and where to dig. :) > From dujinfang at gmail.com Tue Jan 22 18:53:01 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 22 Jan 2013 23:53:01 +0800 Subject: [Freeswitch-users] Reload mod_lua In-Reply-To: <50FE4260.3010202@gmail.com> References: <50FD7E4F.6080607@gmail.com> <50FDAFAE.8080608@gmail.com> <50FDB776.3060100@quentustech.com> <50FE4260.3010202@gmail.com> Message-ID: <0C72D54DCF4C4F2EA39E6B7D6DB1C752@gmail.com> mod_lua was reloadable but when I found that long-running scripts cannot be easily stopped the developers changed it to un-reloadable. I'd find some way to start/stop long-running scripts and I'd seen some implementations based on my example: http://fisheye.freeswitch.org/browse/freeswitch-contrib.git/seven/lua/gateway_report.lua?hb=true There's also a blog page in Chinese explaining this: http://www.freeswitch.org.cn/2010/03/15/zai-freeswitchzhong-zhi-xing-chang-qi-yun-xing-de-qian-ru-shi-jiao-ben-luayu-yan-li-zi.html Hope this helps. On Tuesday, January 22, 2013 at 3:40 PM, Mimiko wrote: > On 21.01.2013 23:47, William King wrote: > > Interesting. A mod_lua api that would list the currently running lua > > scripts with an id that could be used to kill off individual scripts. > > > > > Lua scripts, started with FS on start-up must not be killed, but updated > when changing code of script. A killing command for a lua script is ok > for scripts that got in loop/. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/a0bb2ad7/attachment.html From steveayre at gmail.com Tue Jan 22 19:36:21 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 22 Jan 2013 16:36:21 +0000 Subject: [Freeswitch-users] Reload mod_lua In-Reply-To: <0C72D54DCF4C4F2EA39E6B7D6DB1C752@gmail.com> References: <50FD7E4F.6080607@gmail.com> <50FDAFAE.8080608@gmail.com> <50FDB776.3060100@quentustech.com> <50FE4260.3010202@gmail.com> <0C72D54DCF4C4F2EA39E6B7D6DB1C752@gmail.com> Message-ID: The reason being if the module unloads while scripts are still running you're going to be almost guaranteed a segmentation fault - a well meaning admin running a simple command could crash the entire server dropping all calls. -Steve On 22 January 2013 15:53, Seven Du wrote: > mod_lua was reloadable but when I found that long-running scripts cannot > be easily stopped the developers changed it to un-reloadable. > > I'd find some way to start/stop long-running scripts and I'd seen some > implementations based on my example: > > > http://fisheye.freeswitch.org/browse/freeswitch-contrib.git/seven/lua/gateway_report.lua?hb=true > > > There's also a blog page in Chinese explaining this: > http://www.freeswitch.org.cn/2010/03/15/zai-freeswitchzhong-zhi-xing-chang-qi-yun-xing-de-qian-ru-shi-jiao-ben-luayu-yan-li-zi.html > > > Hope this helps. > > > On Tuesday, January 22, 2013 at 3:40 PM, Mimiko wrote: > > On 21.01.2013 23:47, William King wrote: > > Interesting. A mod_lua api that would list the currently running lua > scripts with an id that could be used to kill off individual scripts. > > > Lua scripts, started with FS on start-up must not be killed, but updated > when changing code of script. A killing command for a lua script is ok > for scripts that got in loop/. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/de74a159/attachment-0001.html From steveayre at gmail.com Tue Jan 22 19:37:36 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 22 Jan 2013 16:37:36 +0000 Subject: [Freeswitch-users] Can a stalled ESL receiver stall FreeSwitch? In-Reply-To: References: Message-ID: This will depend on your version. Older versions could if you failed to read events quickly enough yes. Recent ones do not. Make sure you're running at least v1.2 -Steve On 22 January 2013 15:05, Yehavi Bourvine wrote: > Hello, > > Suppose I have a process that connects to Freeswitch via ESL and asks > for various high verbosity events. Suppose it does not read the data from > the socket - can this stall Freeswitch? > > Thanks, __Yehavi: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/7e29e631/attachment.html From rostotskiy.a at gmail.com Tue Jan 22 19:07:39 2013 From: rostotskiy.a at gmail.com (Anton Rostotskiy) Date: Tue, 22 Jan 2013 18:07:39 +0200 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> Message-ID: Also tried to install mod_opal and also failed. CentOS 6.3 x86_64 ptlib and opal are the last version obtained from svn as it is said in wiki . but just a bunch of errors after make mod_opal http://pastebin.com/HHUPLCLV 2013/1/18 Steven Ayre > The latest version of opal in Debian (even in Sid) isn't new enough for > mod_opal. It needed various features added that are only only available in > the svn version. > > The wiki page states you have to install ptlib and opal from svn - this is > why. > http://wiki.freeswitch.org/wiki/Mod_opal > > Of course that can then cause conflicts unless you package it yourself or > install a dummy package, or problems with other libraries/apps that are > built against the libopal that's already packaged. > > Ptlib and h323plus/opal are also notorious for having to get matching > versions. If you don't pair exactly the right version of ptlib with the > specific version of h323plus/opal you can get instability problems. So the > official Debian ptlib package probably wouldn't be good with opal from > svn... > > As for mod_h323, it does compile and work, at least on Wheezy and possibly > Squeeze too. I've never had any luck getting mod_opal working but have got > mod_h323 working. I don't actively use it though - all my traffic is > currently SIP. > > Another option is to run Yate as a SIP->H323 proxy. It only handles the > signalling, media still goes directly caller->FS. It worked quite well, > although it would occasionally crash - but at least that only dropped the > H323 calls and didn't take FS down with it, and Yate automatically > restarted. That was on Lenny so a long time ago... > > Steve on iPhone > > > > On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote: > > Hi, > > I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and opal > 3.10.7 on a debian > Ptlib and opal get build with no error, but when trying mod_opal I get > > Compiling > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... > quiet_libtool: compile: g++ -g -ggdb > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src > -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE > -DP_64BIT > -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions > -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL > -DHAVE_CONFIG_H -c > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC > -DPIC -o .libs/mod_opal.o > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In > member > function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, > switch_frame_t**, switch_io_flag_t)': > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: > error: 'class OpalCall' has no member named 'IsSwitchingT38' > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In > member > function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, > const switch_frame_t*, switch_io_flag_t)': > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: > error: 'class OpalCall' has no member named 'IsSwitchingT38' > make[3]: *** [mod_opal.lo] Error 1 > make[2]: *** [install] Error 1 > make[1]: *** [mod_opal-install] Error 1 > make: *** [mod_opal-install] Error 2 > > > Has anyone succeded in building mod_opal ? > > Should I go with mod_h323 instead ? > > Thanks for your help > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ? ?????????, ????? ????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/74046ebb/attachment.html From steveayre at gmail.com Tue Jan 22 20:16:22 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 22 Jan 2013 17:16:22 +0000 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> Message-ID: <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> Run: find / -name local.h It's failing to include opal/local.h. All the other errors are because its missing the definitions in that file. Normally I would say to instal the -devel package, but since you built from svn that doesn't apply. Perhaps it's not installed or installed to the wrong place. The find command will tell us where that file is. Steve on iPhone On 22 Jan 2013, at 16:07, Anton Rostotskiy wrote: > Also tried to install mod_opal and also failed. > CentOS 6.3 x86_64 > ptlib and opal are the last version obtained from svn as it is said in wiki. > but just a bunch of errors after make mod_opal http://pastebin.com/HHUPLCLV > > 2013/1/18 Steven Ayre >> The latest version of opal in Debian (even in Sid) isn't new enough for mod_opal. It needed various features added that are only only available in the svn version. >> >> The wiki page states you have to install ptlib and opal from svn - this is why. >> http://wiki.freeswitch.org/wiki/Mod_opal >> >> Of course that can then cause conflicts unless you package it yourself or install a dummy package, or problems with other libraries/apps that are built against the libopal that's already packaged. >> >> Ptlib and h323plus/opal are also notorious for having to get matching versions. If you don't pair exactly the right version of ptlib with the specific version of h323plus/opal you can get instability problems. So the official Debian ptlib package probably wouldn't be good with opal from svn... >> >> As for mod_h323, it does compile and work, at least on Wheezy and possibly Squeeze too. I've never had any luck getting mod_opal working but have got mod_h323 working. I don't actively use it though - all my traffic is currently SIP. >> >> Another option is to run Yate as a SIP->H323 proxy. It only handles the signalling, media still goes directly caller->FS. It worked quite well, although it would occasionally crash - but at least that only dropped the H323 calls and didn't take FS down with it, and Yate automatically restarted. That was on Lenny so a long time ago... >> >> Steve on iPhone >> >> >> >> On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote: >> >>> Hi, >>> >>> I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and opal >>> 3.10.7 on a debian >>> Ptlib and opal get build with no error, but when trying mod_opal I get >>> >>> Compiling >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... >>> quiet_libtool: compile: g++ -g -ggdb >>> -I/usr/local/src/freeswitch/libs/curl/include >>> -I/usr/local/src/freeswitch/src/include >>> -I/usr/local/src/freeswitch/src/include >>> -I/usr/local/src/freeswitch/libs/libteletone/src >>> -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DP_64BIT >>> -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions >>> -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL >>> -DHAVE_CONFIG_H -c >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC >>> -DPIC -o .libs/mod_opal.o >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member >>> function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, >>> switch_frame_t**, switch_io_flag_t)': >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: >>> error: 'class OpalCall' has no member named 'IsSwitchingT38' >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member >>> function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, >>> const switch_frame_t*, switch_io_flag_t)': >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: >>> error: 'class OpalCall' has no member named 'IsSwitchingT38' >>> make[3]: *** [mod_opal.lo] Error 1 >>> make[2]: *** [install] Error 1 >>> make[1]: *** [mod_opal-install] Error 1 >>> make: *** [mod_opal-install] Error 2 >>> >>> >>> Has anyone succeded in building mod_opal ? >>> >>> Should I go with mod_h323 instead ? >>> >>> Thanks for your help >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > ? ?????????, ????? ????????? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/f4f24217/attachment-0001.html From Hector.Geraldino at ipsoft.com Tue Jan 22 21:06:33 2013 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Tue, 22 Jan 2013 18:06:33 +0000 Subject: [Freeswitch-users] Make a call with Perl In-Reply-To: <50FE5A0B.1060207@iway.ch> References: <50FD5C8B.9030306@iway.ch> <50FE5A0B.1060207@iway.ch> Message-ID: I think you need to &park() the call and then send any other commands you want. If you do an originate + playback what will happen is that, once the call is answered, the playback is executed and then there's nothing else on the 'dialplan', so call gets dropped. Do an originate + park (or just originate), then using the uuid do an uuid_playback. The while/loop will also be executed. Finally, you have the option to use an api call (instead of a bgapi) so you don't move to the next action until the call is actually answered (blocking mode) -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matthias Cramer Sent: Tuesday, January 22, 2013 4:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Make a call with Perl Hi Michael On 21/01/13 20:50, Michael Collins wrote: > If you have the FS Cookbook then check out chapter 4, page 81. If you don't have the FS Cookbook then *SHAME* on you! :D I've ordered the book, but it has not yet arrived... :-( > Tip: you can download the examples from both book at the Packt Publishing website. The only caveat as far as I can tell is that you need to create a login account. You do not need a login... Could download them without one. > Grab the examples for the Cookbook and look at chapter 4. The recipe is called "Launching a call with an inbound event socket connect" and it's written in - you guessed it - Perl. :) I started exactly with that example, I do an originate with &playback(). But I do not have real control over the call. I can not play anything else in the audio stream. And if the playback is finished the call hangs up ... Attached is the script I came up with so far ... Regards Matthias -- Matthias Cramer / mc322-ripe Senior Network & Security Engineer iway AG Phone +41 43 500 1111 Badenerstrasse 569 Fax +41 44 271 3535 CH-8048 Z?rich http://www.iway.ch/ GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250 From steveayre at gmail.com Tue Jan 22 21:46:56 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 22 Jan 2013 18:46:56 +0000 Subject: [Freeswitch-users] Make a call with Perl In-Reply-To: References: <50FD5C8B.9030306@iway.ch> Message-ID: Correctly. This is the general usage for any ESL connection, regardless of the client's language. Setting your own UUID is optional, but does avoid a race condition where you might miss events for the call before you manage to subscribe for them. Just another note: use the originate API and execute it with the bgapi ESL command - the command will return instantly and let you collect the events before the orginate succeeds/fails. Also connect in async mode. If you're initiating multiple calls I also highly recommend you write your system in such a way that worker script(s) connect via ESL and keep persistent connections open. Originate multiple calls through the same socket. It'll result in fewer connections open which'll be better load-wise, and since connections will already be open will save a few ms on starting the call up. On the client it means fewer processes running, so fewer context-switches therefore will scale better client-side as well as server-side. And of course there's a limit on the number of processes too. -Steve On 21 January 2013 22:17, Hector Geraldino wrote: > I'm not a perl expert, but in general I think you need to follow this > pattern: > > 1) Establish the connection to FreeSWITCH, you'll be using the event > socket on inbound mode > 2) Start listening for the events you want to capture (send the 'event > plain all' or 'event plain CHANNEL_ANSWER DTMF CHANNEL_HANGUP ...' command) > 3) Create an uuid (by sending the create_uuid command) so you can identify > which call you'll be listening events for > 4) Add a filter for this uuid, so you'll only receive events for this call > over this connection (filter Unique-ID + uuid) > 5) originate the call: {origination_uuid= + uuid}sofia/internal/NXX at GATEWAY > > Check this simple perl example: > > http://wiki.freeswitch.org/wiki/Event_Socket_Library#Simple_Perl_Example > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matthias > Cramer > Sent: Monday, January 21, 2013 10:20 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Make a call with Perl > > Hi All > > I try to make a Perl Script which calls a certain external Phone number. > It then should wait until someone picks up and play a sound file The user > on the other end then needs to confirm the call with a pin Then another > sound file should be played and the the call should be terminated. > > I tried several ways with ESL and originate. But I allays end up with a > call I have no control over. I can read DTMF but can not hang up the call > or play a second file. > > I think I am going in to a wrong direction. > > Any hints ? > > Regards > > Matthias > > -- > Matthias Cramer / mc322-ripe Senior Network & Security Engineer > iway AG Phone +41 43 500 1111 > Badenerstrasse 569 Fax +41 44 271 3535 > CH-8048 Z?rich http://www.iway.ch/ > GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/153777f7/attachment.html From freeswitch at orresta.no-ip.com Tue Jan 22 21:53:42 2013 From: freeswitch at orresta.no-ip.com (Jakob) Date: Tue, 22 Jan 2013 19:53:42 +0100 Subject: [Freeswitch-users] anybody working on a swedish mod_say? In-Reply-To: References: <50D8D157.7020409@orresta.no-ip.com> <50FDB180.3010404@orresta.no-ip.com> Message-ID: <50FEE036.2080007@orresta.no-ip.com> You are correct, It's a oneliner adding the correct linguistical name for the gender to switch_types.h. I could hack the source of mod_say_sv to use an already existing gender but anyone using the module would have to know the existing gender and not the correct one. Jakob 01/21/13 23:09, Avi Marcus skrev: > There are other mod_say languages that have genders. Yours requires a > patch to the core? > -Avi > > On Mon, Jan 21, 2013 at 11:22 PM, Jakob > wrote: > > Ok, > > I now have a full set of recordings for mod_say_sv and > mod_voicemail. In what format should I submit them? 48k.wav? 32? > 16? ulaw? and to where? > > I also have a working mod_say_sv module written with the english > example as a base. So it is aware of the different encodings and > not hardcoded to .wav files. This is now in production on my box > and have been solid. > > There are however a patch to the FS-core that were needed. Adding > a new gender to make the natural flow of recordings working. And a > few patches to build files. > > I'm still working on getting the full voicemail system all swedish. > > How can I submit this for the benefit of others? > > Regards > Jakob Sundberg > > 12/26/12 19:41, Michael Collins skrev: >> Also, if you get a set of Swedish recordings let us know and >> we'll get them hosted on our files.freeswitch.org >> server. >> -MC >> >> On Mon, Dec 24, 2012 at 2:17 PM, Avi Marcus > > wrote: >> >> I can't find a list on the wiki.. but there doesn't seem to >> be a mod_say_** for swedish prompts. >> see in git source: src/mod/say >> >> If one of the languages is similar in making things >> singular/plural, masculine/femanine you can use that as a >> template. >> >> Or you can create speech phrases for numbers with the logic >> in there, but that's a hack.. >> >> >> -Avi >> >> >> On Tue, Dec 25, 2012 at 12:04 AM, Jakob >> > > wrote: >> >> Hi, >> >> I started working on a project that would include a >> voicemail app. but I >> have trouble finding a swedish implementation of a >> mod_say so that I can >> populate it with swedish prompts. >> >> I already have an working IVR, in swedish but there are >> still a few >> bugs, mainly getting english numbers in my swedish prompts. >> >> Jakob >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/b7d2de6c/attachment-0001.html From benoit.raymond at amiconcept.com Tue Jan 22 22:53:53 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Tue, 22 Jan 2013 14:53:53 -0500 Subject: [Freeswitch-users] Handling call forward activate / deactivate through FreeTDM using Sangoma A200 Message-ID: <046501cdf8da$34f394f0$9edabed0$@amiconcept.com> Hello FS users, I have Freeswitch (1.3.13) using a Sangoma A200 card with 2 FXO. I am using FreeTDM and inbound/outbound calls are going through. One of the POTS line has call forwarding feature so my customer wants to have the ability to dial *72 / *73 and get proper dialtone back from the phone company so he can activate or deactivate CFW. While the call is marked as answer in the log, I don?t hear the dialtone back so unsure if it works or not. This is the dialplan I used: Is there a better way or a good way of doing this? Thanks, Benoit Raymond AMI Concept Inc. Solutions VoIP Affaire / Cr?ation Web Tel: (450) 553-1231 http://www.amiconcept.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/ac17becd/attachment.html From garmt.noname at gmail.com Tue Jan 22 23:31:59 2013 From: garmt.noname at gmail.com (grmt) Date: Tue, 22 Jan 2013 21:31:59 +0100 Subject: [Freeswitch-users] Reload mod_lua In-Reply-To: References: <50FD7E4F.6080607@gmail.com> <50FDAFAE.8080608@gmail.com> <50FDB776.3060100@quentustech.com> <50FE4260.3010202@gmail.com> <0C72D54DCF4C4F2EA39E6B7D6DB1C752@gmail.com> Message-ID: <013901cdf8df$8847ba50$98d72ef0$@gmail.com> Rather than waiting forever using - pop(1) -, you can set a timeout value: pop(1, 500). This allows you to let the lua script do some bookkeeping: check for a hash value, file existence, sql select etc. and the script can decide to end itself, without having to wait for a (specific) event. Additionally, this will allow lua to do some bookkeeping. If you use a lua script that runs forever while listening for events, the script may eat your memory, as lua might never get a chance to do proper garbage collection. Make sure you do not subscribe to all events in your lua script, processing events in lua takes time, just subscribe to the events you need. On 22 January 2013 15:53, Seven Du wrote: mod_lua was reloadable but when I found that long-running scripts cannot be easily stopped the developers changed it to un-reloadable. I'd find some way to start/stop long-running scripts and I'd seen some implementations based on my example: http://fisheye.freeswitch.org/browse/freeswitch-contrib.git/seven/lua/gatewa y_report.lua?hb=true There's also a blog page in Chinese explaining this: http://www.freeswitch.org.cn/2010/03/15/zai-freeswitchzhong-zhi-xing-chang-q i-yun-xing-de-qian-ru-shi-jiao-ben-luayu-yan-li-zi.html Hope this helps. On Tuesday, January 22, 2013 at 3:40 PM, Mimiko wrote: On 21.01.2013 23:47, William King wrote: Interesting. A mod_lua api that would list the currently running lua scripts with an id that could be used to kill off individual scripts. Lua scripts, started with FS on start-up must not be killed, but updated when changing code of script. A killing command for a lua script is ok for scripts that got in loop/. -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/194296d3/attachment.html From freeswitch at orresta.no-ip.com Tue Jan 22 23:35:45 2013 From: freeswitch at orresta.no-ip.com (Jakob) Date: Tue, 22 Jan 2013 21:35:45 +0100 Subject: [Freeswitch-users] anybody working on a swedish mod_say? In-Reply-To: References: <50D8D157.7020409@orresta.no-ip.com> <50FDB180.3010404@orresta.no-ip.com> Message-ID: <50FEF821.8040308@orresta.no-ip.com> I added this to jira http://jira.freeswitch.org/browse/FS-5045 it's the mod_say_sv source, config files for starting the module and loading the vm voicemacros. Are there any documentation on what extra files are needed or if I missed something? Jakob Ps. Recordings will follow, need to clean up the audio levels. Ds. 01/21/13 22:41, Michael Jerris skrev: > Lets start with patches for all the code changes on jira. We also > will need a full set of phrase macros for swedish (again patches). > For the sound files we typically start with 48k and resample down, > for those coordinate with Michael Collins. > > Mike > > On Jan 21, 2013, at 4:22 PM, Jakob > wrote: > >> Ok, >> >> I now have a full set of recordings for mod_say_sv and mod_voicemail. >> In what format should I submit them? 48k.wav? 32? 16? ulaw? and to where? >> >> I also have a working mod_say_sv module written with the english >> example as a base. So it is aware of the different encodings and not >> hardcoded to .wav files. This is now in production on my box and have >> been solid. >> >> There are however a patch to the FS-core that were needed. Adding a >> new gender to make the natural flow of recordings working. And a few >> patches to build files. >> >> I'm still working on getting the full voicemail system all swedish. >> >> How can I submit this for the benefit of others? >> >> Regards >> Jakob Sundberg >> >> 12/26/12 19:41, Michael Collins skrev: >>> Also, if you get a set of Swedish recordings let us know and we'll >>> get them hosted on our files.freeswitch.org >>> server. >>> -MC >>> >>> On Mon, Dec 24, 2012 at 2:17 PM, Avi Marcus >> > wrote: >>> >>> I can't find a list on the wiki.. but there doesn't seem to be a >>> mod_say_** for swedish prompts. >>> see in git source: src/mod/say >>> >>> If one of the languages is similar in making things >>> singular/plural, masculine/femanine you can use that as a template. >>> >>> Or you can create speech phrases for numbers with the logic in >>> there, but that's a hack.. >>> >>> >>> -Avi >>> >>> >>> On Tue, Dec 25, 2012 at 12:04 AM, Jakob >>> >> > wrote: >>> >>> Hi, >>> >>> I started working on a project that would include a >>> voicemail app. but I >>> have trouble finding a swedish implementation of a mod_say >>> so that I can >>> populate it with swedish prompts. >>> >>> I already have an working IVR, in swedish but there are >>> still a few >>> bugs, mainly getting english numbers in my swedish prompts. >>> >>> Jakob >>> >>> _________ >>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/e54ee5df/attachment-0001.html From jtbock at synacktics.com Wed Jan 23 00:09:21 2013 From: jtbock at synacktics.com (Tim Bock) Date: Tue, 22 Jan 2013 14:09:21 -0700 Subject: [Freeswitch-users] How to enable debug mode for DTMF in FreeTDM? In-Reply-To: <50FE43A3.2070104@tangramltd.com> References: <50FD8C6D.3090806@tangramltd.com> <50FE43A3.2070104@tangramltd.com> Message-ID: <50FF0001.8000606@synacktics.com> Hi, That looks about right. The box that I did this on isn't available to me at the moment, but I was successful in getting a file created. I'll check tomorrow when I will have access to the server again. One simple thing would be to check permissions, to make sure that the freeswitch user has write permission to that directory. Tim On 01/22/2013 12:45 AM, Viacheslav Dubrovskyi wrote: > 21.01.2013 20:43, Viacheslav Dubrovskyi ?????: >> Hi. >> >> How to enable debug mode for DTMF in FreeTDM? >> >> I setup in freetdm.conf >> >> [general] >> debugdtmf_directory=/var/spool/freeswitch/dtmf >> >> [span zt mySpan] >> debugdtmf => yes >> >> >> But there is nothing in path. What else should I do? > All ignore me ;( > Or is it a secret knowledge available only to the initiated? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Tim Bock Synacktics, LLC www.synacktics.com (505)795-1511 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/6f63b645/attachment.html From jtbock at synacktics.com Wed Jan 23 00:20:28 2013 From: jtbock at synacktics.com (Tim Bock) Date: Tue, 22 Jan 2013 14:20:28 -0700 Subject: [Freeswitch-users] dtmf problem In-Reply-To: <50FE58F4.2010805@tangramltd.com> References: <50F38BEE.6090804@synacktics.com> <50FE58F4.2010805@tangramltd.com> Message-ID: <50FF029C.2040900@synacktics.com> Hi, Mine was a combination of things. One problem seemed to be as I described in an earlier post, that something (dahdi?) was not happy if I disconnected the trunk lines. This manifested as the dtmf problem and echo cancelling problem (and "cat /proc/dahdi/1" would show oslec as INACTIVE). Even after figuring out that shutting down the machine and restarting helped, I still had an occasional echo problem even though "cat /prod/dahdi/1" showed that oslec was active. I eventually fixed that by increasing the taps to 128 in zt.conf. As you can no doubt tell from my posts, I am a relative noob to freeswitch, but these are things which worked for me. I did a previous installation which was sip only (no analog card), and that was much easier! Best, Tim On 01/22/2013 02:16 AM, Viacheslav Dubrovskyi wrote: > 14.01.2013 06:39, Tim Bock ?????: >> Hello List, >> >> I have a freeswitch server with a digium tdm wildcard. Compiled, >> enabled, and configured freetdm and dahdi...all seems to be working fine >> (can make/receive calls). >> >> My problem is with dtmf. When calling from an external pstn number to >> freeswitch, dtmf seems to be detected properly and consistently. But >> when calling from a cell phone or via another voip line, dtmf doesn't >> seem to be detected, or detected only occasionally. >> >> On the same system, if I call into the voip number, calls from any of >> the three external options (landline, cell, sip) all seem to work fine >> with dtmf. >> >> So the problem seems to be something I've misconfigured with the tdm >> card. Can anyone point me in the right direction? > Found this post > http://fonality.com/trixbox/forums/trixbox-forums/help/inbound-dtmf-pri-t-analog > > It seems that explains. But I not sure. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Tim Bock Synacktics, LLC www.synacktics.com (505)795-1511 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/670a2b21/attachment.html From msc at freeswitch.org Wed Jan 23 01:24:01 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Jan 2013 14:24:01 -0800 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: <50FE1857.7020804@communicatefreely.net> References: <50FE1857.7020804@communicatefreely.net> Message-ID: That begs the question: does anyone know of any SIP phones that have an API where you can poll the status of the lamps as well as what's on the display, whether MWI is on, etc.? That would be epic. -MC On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre < fs-list at communicatefreely.net> wrote: > Hmm. Good idea. > > I know that you can do this with Aastra phones, and possibly others. > > With the Aastra's, there is an XML API that you can use to push > notifications, apps, etc. > You can push XML to the phone on an HTTP interface (there is an ACL to > tell it where to > allow it from), and you can initiate phone calls, as well as hang up, > transfer, etc. > There is also an action URI, so you can have the phone report what is > coming in on caller > ID, when it is ringing, hung up, etc. > > I don't know if you can get lamp status from this or not, but perhaps you > could have a > system where a pattern of calls is generated, and after many repetitions, > the system calls > or a human to check and see that the lights match an expected state. > > Just an idea anyway. > > -Tim > > Michael Collins wrote: > > This test suite does not presently exist. I would love to see some work > > done in this area. > > > > One challenge is that there are numerous kinds of devices that can > > interop with FS so the test suite will need to have some way to handle > > the various phone makes and models. Also, I don't know which hard > > phones, if any, support automated testing. I would *love* to see some > > sort of web/network API on various phones where you could poll the phone > > to see the state of each SLA/BLF key as well as what's in the display. > > That would make automated testing much cleaner. As it stands now, > > testing BLFs is a tedious process of watching phones and waiting to see > > if the lights do what they should be doing. > > > > If anyone has practical suggestions (nothing pie-in-the-sky, if you get > > my meaning) then by all means share with the group. > > > > -MC > > > > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine > > > wrote: > > > > Hello, > > > > We are running version 1.0.6; we did not upgrade for a long time > > due to some problems (mainly stuck BLFs) with the newer versions > > which have been corrected a few months ago. > > > > A few weeks ago I pulled the latest GIT and did functional testings > > on our development system; all was OK. > > > > Yesterday I moved our production system to that version. After a few > > hours the problems started: stuck BLFs, hangups, LUA scripts > > sometimes do not run. I had to revert back to the old version. > > > > My question: Is there some simple test suite which I can test > > various scnarios on my development system? like, "call XX, while > > this call is active call YY which is monitored by XX and hangup", > > etc. It is hard to cover all cases manually, so I would like > > something automated. > > > > Thanks, __Yehavi: > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > ------------------------------------------------------------------------ > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/55858812/attachment-0001.html From msc at freeswitch.org Wed Jan 23 01:26:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Jan 2013 14:26:30 -0800 Subject: [Freeswitch-users] anybody working on a swedish mod_say? In-Reply-To: <50FEF821.8040308@orresta.no-ip.com> References: <50D8D157.7020409@orresta.no-ip.com> <50FDB180.3010404@orresta.no-ip.com> <50FEF821.8040308@orresta.no-ip.com> Message-ID: This is a classic example of something that bug marshals and other volunteers can do: apply this patch and try it out. Of course, there are no sound files yet but we are working on that. Thanks, MC On Tue, Jan 22, 2013 at 12:35 PM, Jakob wrote: > I added this to jira > > http://jira.freeswitch.org/browse/FS-5045 > > it's the mod_say_sv source, config files for starting the module and > loading the vm voicemacros. > > Are there any documentation on what extra files are needed or if I missed > something? > > Jakob > Ps. Recordings will follow, need to clean up the audio levels. Ds. > 01/21/13 22:41, Michael Jerris skrev: > > Lets start with patches for all the code changes on jira. We also will > need a full set of phrase macros for swedish (again patches). For the > sound files we typically start with 48k and resample down, for those > coordinate with Michael Collins. > > Mike > > On Jan 21, 2013, at 4:22 PM, Jakob wrote: > > Ok, > > I now have a full set of recordings for mod_say_sv and mod_voicemail. In > what format should I submit them? 48k.wav? 32? 16? ulaw? and to where? > > I also have a working mod_say_sv module written with the english example > as a base. So it is aware of the different encodings and not hardcoded to > .wav files. This is now in production on my box and have been solid. > > There are however a patch to the FS-core that were needed. Adding a new > gender to make the natural flow of recordings working. And a few patches to > build files. > > I'm still working on getting the full voicemail system all swedish. > > How can I submit this for the benefit of others? > > Regards > Jakob Sundberg > > 12/26/12 19:41, Michael Collins skrev: > > Also, if you get a set of Swedish recordings let us know and we'll get > them hosted on our files.freeswitch.org server. > -MC > > On Mon, Dec 24, 2012 at 2:17 PM, Avi Marcus wrote: > >> I can't find a list on the wiki.. but there doesn't seem to be a >> mod_say_** for swedish prompts. >> see in git source: src/mod/say >> >> If one of the languages is similar in making things singular/plural, >> masculine/femanine you can use that as a template. >> >> Or you can create speech phrases for numbers with the logic in there, >> but that's a hack.. >> >> >> -Avi >> >> >> On Tue, Dec 25, 2012 at 12:04 AM, Jakob wrote: >> >>> Hi, >>> >>> I started working on a project that would include a voicemail app. but I >>> have trouble finding a swedish implementation of a mod_say so that I can >>> populate it with swedish prompts. >>> >>> I already have an working IVR, in swedish but there are still a few >>> bugs, mainly getting english numbers in my swedish prompts. >>> >>> Jakob >>> >>> _________ >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/e15be7dd/attachment.html From msc at freeswitch.org Wed Jan 23 01:32:18 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Jan 2013 14:32:18 -0800 Subject: [Freeswitch-users] Handling call forward activate / deactivate through FreeTDM using Sangoma A200 In-Reply-To: <046501cdf8da$34f394f0$9edabed0$@amiconcept.com> References: <046501cdf8da$34f394f0$9edabed0$@amiconcept.com> Message-ID: Are you sure that the telco sends back dial tone? One thing you can do is use the ftdm-trace command to capture the raw audio coming in on the port. Another thing you can do is just have *72 go to a play_and_get_digits app that plays dialtone to the user and collects digits. (Look in features.xml for an example of this, in the "dx" extension if memory serves.) From there just bridge to *72 + pause + dialed digits from PAGD. For *73 you can just bridge to *73 since you don't normally need to dial any digits. -MC On Tue, Jan 22, 2013 at 11:53 AM, Benoit Raymond < benoit.raymond at amiconcept.com> wrote: > Hello FS users,**** > > ** ** > > I have Freeswitch (1.3.13) using a Sangoma A200 card with 2 FXO. I am > using FreeTDM and inbound/outbound calls are going through. One of the POTS > line has call forwarding feature so my customer wants to have the ability > to dial **72 / **73 and get proper dialtone back from the phone company > so he can activate or deactivate CFW.**** > > ** ** > > While the call is marked as answer in the log, I don?t hear the dialtone > back so unsure if it works or not. This is the dialplan I used:**** > > ** ** > > **** > > *** > * > > data="default_language=${default_language}"/>**** > > **** > > **** > > **** > > **** > > ** ** > > Is there a better way or a good way of doing this?**** > > ** ** > > Thanks,**** > > ** ** > > *Benoit Raymond***** > > AMI Concept Inc.**** > > Solutions VoIP Affaire / Cr?ation Web**** > > Tel: (450) 553-1231**** > > http://www.amiconcept.com**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/93bf62da/attachment.html From slava at tangramltd.com Wed Jan 23 01:33:58 2013 From: slava at tangramltd.com (Dubrovskiy Viacheslav) Date: Wed, 23 Jan 2013 00:33:58 +0200 Subject: [Freeswitch-users] How to enable debug mode for DTMF in FreeTDM? In-Reply-To: <50FF0001.8000606@synacktics.com> References: <50FD8C6D.3090806@tangramltd.com> <50FE43A3.2070104@tangramltd.com> <50FF0001.8000606@synacktics.com> Message-ID: <50FF13D6.2020008@tangramltd.com> 22.01.2013 23:09, Tim Bock ?????: > Hi, > > That looks about right. The box that I did this on isn't available to > me at the moment, but I was successful in getting a file created. > I'll check tomorrow when I will have access to the server again. > > One simple thing would be to check permissions, to make sure that the > freeswitch user has write permission to that directory. He can. I can write trace to this dir: > ftdm trace /var/spool/freeswitch/dtmf/trace 1 1 >>> Hi. >>> >>> How to enable debug mode for DTMF in FreeTDM? >>> >>> I setup in freetdm.conf >>> >>> [general] >>> debugdtmf_directory=/var/spool/freeswitch/dtmf >>> >>> [span zt mySpan] >>> debugdtmf => yes >>> >>> >>> But there is nothing in path. What else should I do? >> All ignore me ;( >> Or is it a secret knowledge available only to the initiated? >> -- WBR, Dubrovskiy Viacheslav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/aa371c61/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3746 bytes Desc: ?????????????????????????????????? ?????????????? S/MIME Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/aa371c61/attachment-0001.bin From schoch+freeswitch.org at xwin32.com Wed Jan 23 01:35:51 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 22 Jan 2013 14:35:51 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: <20130119191943.ca09ab6d@mail.tritonwest.net> References: <20130119191943.ca09ab6d@mail.tritonwest.net> Message-ID: Here's an update on my FAX project: On our main FreeSWITCH pbx (on CentOS), I run this command through the CLI and it works: freeswitch at internal> originate sofia/gateway/flowroute/1408739xxxx &txfax(/home/schoch/fax000007603.tif) I did a basic install on a Windows machine, by downloading the installer, installing it, and changing the "default_provider" lines in vars.xml in order to make it connect to our pbx. A "sofia status" shows it's connected. Then I ran this command on the Windows machine: freeswitch at internal> originate sofia/gateway/external::starnet_fax/1408739xxxx &txfax(C:/Users/Schoch/Pictures/fax000007603.tif) It rang the FAX machine, but the FAX never got there. It got some sort of a timeout: 2013-01-22 14:09:10.224456 [DEBUG] switch_rtp.c:928 [ zrtp engine]: WARNING! HELLO Max retransmissions count reached (20 retries). ID=1 and 2013-01-22 14:10:07.744746 [DEBUG] mod_spandsp_fax.c:500 Fax processing not successful - result (2) Timed out waiting for initial communication. I'm not sure where to go from here. Any ideas? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/88853a63/attachment.html From lists at telefaks.de Wed Jan 23 02:34:41 2013 From: lists at telefaks.de (Peter Steinbach) Date: Wed, 23 Jan 2013 00:34:41 +0100 Subject: [Freeswitch-users] mod_stress In-Reply-To: References: <50FDC789.9000707@telefaks.de> <50FE75CA.2030007@telefaks.de> Message-ID: <50FF2211.1060302@telefaks.de> Thanks Steven, I got it working and reported my howto and findings in the wiki: http://wiki.freeswitch.org/wiki/Mod_stress I found it to be a nice application, but as you explained, there are some false positives. Best regards Peter On 01/22/13 12:39, Steven Ayre wrote: > Media bugs are used to hook into the call and provide decoded audio to > the module. That requires transcoding, naturally. See stress_callback > which is called for each audio frame plus some init/cleanup hooks, and > read up on the API at http://docs.freeswitch.org/group__mb1.html. > > I suspect read vs write would correspond to the speaking/listening > directions of the channel. Read from caller, write to caller? The > names come from the media bug API. The read/write options will add the > corresponding media bug, stop will remove them. > > SWITCH_ADD_APP tells you it's a dialplan application - it's started > from dialplan, although you can also trigger dialplan apps on a > channel via ESL using uuid_broadcast. > > The FFT algorithm is used to extract the frequencies present in the > sample (the power spectrum). The algorithm then tries to detect > stressed based on that. Looks like the algorithm at least in part > attempts to look for tremor in the voice. More tremor indicates more > stress. Over a threshold it raises an event. > > Bear in mind that as it's pattern recognition (and little tested at > that) you'll get a lot of false positives and false negatives. It's > also looking for something in the voice that may or may not be there - > some people might be able to speak with an unstressed voice when > they're actually quite stressed. > > -Steve > > > > > On 22 January 2013 11:19, Peter Steinbach > wrote: > > Hello Anthony, > > although it may be incomplete, it may be useful for us. Can you > give us some basic explanation, how this works? > I can see in the code that there is a stress command with > read|write|stop parameter. > Is this to be issued via event socket or is it a dialplan > application? I suspect that "read" somehow starts the detection > and "stop" somehow stops the detection. For "write" I currently > have no idea. > > I would be happy to test this and report my configuration, > explanations and examples in the wiki. > > Best regards > Peter > > > > On 01/22/13 00:30, Anthony Minessale wrote: >> That module is incomplete we made it several years ago to test >> some code that was supposed to pick up on stress in your voice in >> a media bug so we could fire events and whisper "lie" to the >> caller as he talked to the guy. >> >> >> >> On Mon, Jan 21, 2013 at 4:56 PM, Peter Steinbach >> > wrote: >> >> I am also wondering, what this module does in detail. >> >> Regarding the source code, I can see that FFT's are made in >> order to to >> get some sort of frequency profile. >> But the code and remarks do not give me a clue how this >> module actually >> works. >> >> Best regards Peter >> >> >> >> On 05/26/12 15:05, BF wrote: >> > Hello, >> > >> > I'd like to know what the "stress write" from mod_stress >> module does. I have hard time finding documentation or an >> example about mod_stress module. >> > >> > Anyone? >> > >> > Thank you. >> > >> > Eric >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> -- >> With kind regards >> Peter Steinbach >> >> Telefaks Services GmbH >> mailto:lists (att) telefaks.de >> >> Internet: www.telefaks.de >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/6fde4e07/attachment-0001.html From benoit.raymond at amiconcept.com Wed Jan 23 06:18:09 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Tue, 22 Jan 2013 22:18:09 -0500 Subject: [Freeswitch-users] Retrieving hangupCause from b-leg in mod_lua Message-ID: <04fc01cdf918$4521b260$cf651720$@amiconcept.com> Hello, Is there a way in lua to figure out why the b-leg hung up? I have an inbound call that I need to bridge so when lua is invoked, there is a session created for the inbound call and then I use session:execute("bridge", "the b-leg path") and either the call is bridged or call_timeout kicks in. I need to continue my script based on the hangup cause whether the b-leg produced a NO_ANSWER, USER_BUSY or any other condition. The session:hangupCause is of no use since my session is still active. I found the variable "bridge_hangup_cause" on the wiki but it returns nil if I do session:getVariable("bridge_hangup_cause"). I am using 1.3.13 Thanks much. Benoit Raymond AMI Concept Inc. Business VoIP Solutions / Web Design Phone: (450) 553-1231 http://www.amiconcept.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/a089be82/attachment.html From benoit.raymond at amiconcept.com Wed Jan 23 06:26:00 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Tue, 22 Jan 2013 22:26:00 -0500 Subject: [Freeswitch-users] Handling call forward activate / deactivate through FreeTDM using Sangoma A200 In-Reply-To: References: <046501cdf8da$34f394f0$9edabed0$@amiconcept.com> Message-ID: <050701cdf919$5daf2370$190d6a50$@amiconcept.com> Hey Michael, Yes, telco returns the confirmation tone (two small tones) if we connect a standard phone directly on the line. That?s how my customer is currently activating / deactivating CFW. The solution you provide is nice but is more a workaround than a solution as it could lead to failures if the telco is, for some reason, not returning the confirmation tone. That would create confusion at my customer. Nevertheless, you brought a good idea as I could use mod_spandsp to detect those tones and then offer such tone back to my customer. I?ll pursue this path once I get Sangoma to explain why it takes almost 10 seconds to connect the FXO port on the POTS line and sends the digits. Thanks, Benoit Raymond AMI Concept Inc. Business VoIP Solutions / Web Design Phone: (450) 553-1231 http://www.amiconcept.com De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : 22 janvier 2013 17:32 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Handling call forward activate / deactivate through FreeTDM using Sangoma A200 Are you sure that the telco sends back dial tone? One thing you can do is use the ftdm-trace command to capture the raw audio coming in on the port. Another thing you can do is just have *72 go to a play_and_get_digits app that plays dialtone to the user and collects digits. (Look in features.xml for an example of this, in the "dx" extension if memory serves.) From there just bridge to *72 + pause + dialed digits from PAGD. For *73 you can just bridge to *73 since you don't normally need to dial any digits. -MC On Tue, Jan 22, 2013 at 11:53 AM, Benoit Raymond wrote: Hello FS users, I have Freeswitch (1.3.13) using a Sangoma A200 card with 2 FXO. I am using FreeTDM and inbound/outbound calls are going through. One of the POTS line has call forwarding feature so my customer wants to have the ability to dial *72 / *73 and get proper dialtone back from the phone company so he can activate or deactivate CFW. While the call is marked as answer in the log, I don?t hear the dialtone back so unsure if it works or not. This is the dialplan I used: Is there a better way or a good way of doing this? Thanks, Benoit Raymond AMI Concept Inc. Solutions VoIP Affaire / Cr?ation Web Tel: (450) 553-1231 http://www.amiconcept.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130122/1e563add/attachment.html From jaganthoutam at gmail.com Wed Jan 23 08:52:42 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Wed, 23 Jan 2013 11:22:42 +0530 Subject: [Freeswitch-users] Make a call with Perl In-Reply-To: References: <50FD5C8B.9030306@iway.ch> Message-ID: if we have limitation on socket it will be the issue while Originating Bulk calls, so in my point there is no need collecting the Information in Single script. if we use 2 scripts for Bulk calls one is for just Origination not collecting any event from ESL, One script will taking care of IVR or what ever we want to perform. This is Working me for. On 23 January 2013 00:16, Steven Ayre wrote: > Correctly. This is the general usage for any ESL connection, regardless of > the client's language. > > Setting your own UUID is optional, but does avoid a race condition where you > might miss events for the call before you manage to subscribe for them. > > Just another note: use the originate API and execute it with the bgapi ESL > command - the command will return instantly and let you collect the events > before the orginate succeeds/fails. > > Also connect in async mode. > > If you're initiating multiple calls I also highly recommend you write your > system in such a way that worker script(s) connect via ESL and keep > persistent connections open. Originate multiple calls through the same > socket. It'll result in fewer connections open which'll be better load-wise, > and since connections will already be open will save a few ms on starting > the call up. On the client it means fewer processes running, so fewer > context-switches therefore will scale better client-side as well as > server-side. And of course there's a limit on the number of processes too. > > -Steve > > > > On 21 January 2013 22:17, Hector Geraldino > wrote: >> >> I'm not a perl expert, but in general I think you need to follow this >> pattern: >> >> 1) Establish the connection to FreeSWITCH, you'll be using the event >> socket on inbound mode >> 2) Start listening for the events you want to capture (send the 'event >> plain all' or 'event plain CHANNEL_ANSWER DTMF CHANNEL_HANGUP ...' command) >> 3) Create an uuid (by sending the create_uuid command) so you can identify >> which call you'll be listening events for >> 4) Add a filter for this uuid, so you'll only receive events for this call >> over this connection (filter Unique-ID + uuid) >> 5) originate the call: {origination_uuid= + >> uuid}sofia/internal/NXX at GATEWAY >> >> Check this simple perl example: >> >> http://wiki.freeswitch.org/wiki/Event_Socket_Library#Simple_Perl_Example >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matthias >> Cramer >> Sent: Monday, January 21, 2013 10:20 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Make a call with Perl >> >> Hi All >> >> I try to make a Perl Script which calls a certain external Phone number. >> It then should wait until someone picks up and play a sound file The user >> on the other end then needs to confirm the call with a pin Then another >> sound file should be played and the the call should be terminated. >> >> I tried several ways with ESL and originate. But I allays end up with a >> call I have no control over. I can read DTMF but can not hang up the call or >> play a second file. >> >> I think I am going in to a wrong direction. >> >> Any hints ? >> >> Regards >> >> Matthias >> >> -- >> Matthias Cramer / mc322-ripe Senior Network & Security Engineer >> iway AG Phone +41 43 500 1111 >> Badenerstrasse 569 Fax +41 44 271 3535 >> CH-8048 Z?rich http://www.iway.ch/ >> GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From avi at avimarcus.net Wed Jan 23 10:17:04 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 23 Jan 2013 09:17:04 +0200 Subject: [Freeswitch-users] Retrieving hangupCause from b-leg in mod_lua In-Reply-To: <04fc01cdf918$4521b260$cf651720$@amiconcept.com> References: <04fc01cdf918$4521b260$cf651720$@amiconcept.com> Message-ID: Try: cause = session:getVariable("originate_disposition") cause2=session:getVariable("bridge_hangup_cause") I'm not exactly sure what gets into which variable... But I think I needed both because sometimes "user_busy" was only in the second one. I'm not 100% sure... Here's my production error script here: https://gist.github.com/4602781 (if the user accessed via a the calling card IVR, it passes them back to the menu. Otherwise, it hangs up on them. Remove or keep as appropriate.) Please share your results if you are able to make it cleaner. -Avi Marcus BestFone On Wed, Jan 23, 2013 at 5:18 AM, Benoit Raymond < benoit.raymond at amiconcept.com> wrote: > Hello,**** > > ** ** > > Is there a way in lua to figure out why the b-leg hung up? I have an > inbound call that I need to bridge so when lua is invoked, there is a > session created for the inbound call and then I use > session:execute(?bridge?, ?the b-leg path?) and either the call is bridged > or call_timeout kicks in. I need to continue my script based on the hangup > cause whether the b-leg produced a NO_ANSWER, USER_BUSY or any other > condition. The session:hangupCause is of no use since my session is still > active.**** > > ** ** > > I found the variable ?bridge_hangup_cause? on the wiki but it returns nil > if I do session:getVariable(?bridge_hangup_cause?).**** > > ** ** > > I am using 1.3.13**** > > ** ** > > Thanks much.**** > > ** ** > > *Benoit Raymond***** > > AMI Concept Inc.**** > > Business VoIP Solutions / Web Design**** > > Phone: (450) 553-1231**** > > http://www.amiconcept.com**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/bcd32f2a/attachment.html From rostotskiy.a at gmail.com Wed Jan 23 11:16:41 2013 From: rostotskiy.a at gmail.com (Anton Rostotskiy) Date: Wed, 23 Jan 2013 10:16:41 +0200 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> Message-ID: Thanks for reply, Steven! Tried to search local.h file, but search returned nothing. That are all local.h files in my system: /usr/local/src/freeswitch/libs/spandsp/src/gsm0610_local.h /usr/local/src/freeswitch/libs/spandsp/src/t30_local.h /usr/local/src/freeswitch/libs/spandsp/src/t42_t43_local.h About -devel packages - at first tried to install ptlib, opal and -devel from yum but their versions have been too low for mod_opal. 2013/1/22 Steven Ayre > Run: > find / -name local.h > > It's failing to include opal/local.h. All the other errors are because its > missing the definitions in that file. > > Normally I would say to instal the -devel package, but since you built > from svn that doesn't apply. Perhaps it's not installed or installed to the > wrong place. The find command will tell us where that file is. > > Steve on iPhone > > On 22 Jan 2013, at 16:07, Anton Rostotskiy wrote: > > Also tried to install mod_opal and also failed. > CentOS 6.3 x86_64 > ptlib and opal are the last version obtained from svn as it is said in > wiki . > but just a bunch of errors after make mod_opal > http://pastebin.com/HHUPLCLV > > 2013/1/18 Steven Ayre > >> The latest version of opal in Debian (even in Sid) isn't new enough for >> mod_opal. It needed various features added that are only only available in >> the svn version. >> >> The wiki page states you have to install ptlib and opal from svn - this >> is why. >> http://wiki.freeswitch.org/wiki/Mod_opal >> >> Of course that can then cause conflicts unless you package it yourself or >> install a dummy package, or problems with other libraries/apps that are >> built against the libopal that's already packaged. >> >> Ptlib and h323plus/opal are also notorious for having to get matching >> versions. If you don't pair exactly the right version of ptlib with the >> specific version of h323plus/opal you can get instability problems. So the >> official Debian ptlib package probably wouldn't be good with opal from >> svn... >> >> As for mod_h323, it does compile and work, at least on Wheezy and >> possibly Squeeze too. I've never had any luck getting mod_opal working but >> have got mod_h323 working. I don't actively use it though - all my traffic >> is currently SIP. >> >> Another option is to run Yate as a SIP->H323 proxy. It only handles the >> signalling, media still goes directly caller->FS. It worked quite well, >> although it would occasionally crash - but at least that only dropped the >> H323 calls and didn't take FS down with it, and Yate automatically >> restarted. That was on Lenny so a long time ago... >> >> Steve on iPhone >> >> >> >> On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote: >> >> Hi, >> >> I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and >> opal >> 3.10.7 on a debian >> Ptlib and opal get build with no error, but when trying mod_opal I get >> >> Compiling >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... >> quiet_libtool: compile: g++ -g -ggdb >> -I/usr/local/src/freeswitch/libs/curl/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/libs/libteletone/src >> -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden >> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE >> -DP_64BIT >> -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions >> -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL >> -DHAVE_CONFIG_H -c >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC >> -DPIC -o .libs/mod_opal.o >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In >> member >> function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, >> switch_frame_t**, switch_io_flag_t)': >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: >> error: 'class OpalCall' has no member named 'IsSwitchingT38' >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In >> member >> function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, >> const switch_frame_t*, switch_io_flag_t)': >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: >> error: 'class OpalCall' has no member named 'IsSwitchingT38' >> make[3]: *** [mod_opal.lo] Error 1 >> make[2]: *** [install] Error 1 >> make[1]: *** [mod_opal-install] Error 1 >> make: *** [mod_opal-install] Error 2 >> >> >> Has anyone succeded in building mod_opal ? >> >> Should I go with mod_h323 instead ? >> >> Thanks for your help >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > ? ?????????, ????? ????????? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ? ?????????, ????? ????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/ecb6cea3/attachment-0001.html From steveayre at gmail.com Wed Jan 23 12:17:48 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 23 Jan 2013 09:17:48 +0000 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> Message-ID: It's localep.h, not local.h: /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.h:41:26: error: opal/localep.h: No such file or directory find / -name localep.h Sorry, seems my phone did an annoying autouncorrection! -Steve On 23 January 2013 08:16, Anton Rostotskiy wrote: > Thanks for reply, Steven! > Tried to search local.h file, but search returned nothing. That are all > local.h files in my system: > > /usr/local/src/freeswitch/libs/spandsp/src/gsm0610_local.h > /usr/local/src/freeswitch/libs/spandsp/src/t30_local.h > /usr/local/src/freeswitch/libs/spandsp/src/t42_t43_local.h > > About -devel packages - at first tried to install ptlib, opal and -devel from > yum but their versions have been too low for mod_opal. > > 2013/1/22 Steven Ayre > >> Run: >> find / -name local.h >> >> It's failing to include opal/local.h. All the other errors are because >> its missing the definitions in that file. >> >> Normally I would say to instal the -devel package, but since you built >> from svn that doesn't apply. Perhaps it's not installed or installed to the >> wrong place. The find command will tell us where that file is. >> >> Steve on iPhone >> >> On 22 Jan 2013, at 16:07, Anton Rostotskiy >> wrote: >> >> Also tried to install mod_opal and also failed. >> CentOS 6.3 x86_64 >> ptlib and opal are the last version obtained from svn as it is said in >> wiki . >> but just a bunch of errors after make mod_opal >> http://pastebin.com/HHUPLCLV >> >> 2013/1/18 Steven Ayre >> >>> The latest version of opal in Debian (even in Sid) isn't new enough for >>> mod_opal. It needed various features added that are only only available in >>> the svn version. >>> >>> The wiki page states you have to install ptlib and opal from svn - this >>> is why. >>> http://wiki.freeswitch.org/wiki/Mod_opal >>> >>> Of course that can then cause conflicts unless you package it yourself >>> or install a dummy package, or problems with other libraries/apps that are >>> built against the libopal that's already packaged. >>> >>> Ptlib and h323plus/opal are also notorious for having to get matching >>> versions. If you don't pair exactly the right version of ptlib with the >>> specific version of h323plus/opal you can get instability problems. So the >>> official Debian ptlib package probably wouldn't be good with opal from >>> svn... >>> >>> As for mod_h323, it does compile and work, at least on Wheezy and >>> possibly Squeeze too. I've never had any luck getting mod_opal working but >>> have got mod_h323 working. I don't actively use it though - all my traffic >>> is currently SIP. >>> >>> Another option is to run Yate as a SIP->H323 proxy. It only handles the >>> signalling, media still goes directly caller->FS. It worked quite well, >>> although it would occasionally crash - but at least that only dropped the >>> H323 calls and didn't take FS down with it, and Yate automatically >>> restarted. That was on Lenny so a long time ago... >>> >>> Steve on iPhone >>> >>> >>> >>> On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote: >>> >>> Hi, >>> >>> I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and >>> opal >>> 3.10.7 on a debian >>> Ptlib and opal get build with no error, but when trying mod_opal I get >>> >>> Compiling >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... >>> quiet_libtool: compile: g++ -g -ggdb >>> -I/usr/local/src/freeswitch/libs/curl/include >>> -I/usr/local/src/freeswitch/src/include >>> -I/usr/local/src/freeswitch/src/include >>> -I/usr/local/src/freeswitch/libs/libteletone/src >>> -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE >>> -DP_64BIT >>> -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions >>> -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL >>> -DHAVE_CONFIG_H -c >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC >>> -DPIC -o .libs/mod_opal.o >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In >>> member >>> function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, >>> switch_frame_t**, switch_io_flag_t)': >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: >>> error: 'class OpalCall' has no member named 'IsSwitchingT38' >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In >>> member >>> function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, >>> const switch_frame_t*, switch_io_flag_t)': >>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: >>> error: 'class OpalCall' has no member named 'IsSwitchingT38' >>> make[3]: *** [mod_opal.lo] Error 1 >>> make[2]: *** [install] Error 1 >>> make[1]: *** [mod_opal-install] Error 1 >>> make: *** [mod_opal-install] Error 2 >>> >>> >>> Has anyone succeded in building mod_opal ? >>> >>> Should I go with mod_h323 instead ? >>> >>> Thanks for your help >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> ? ?????????, ????? ????????? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > ? ?????????, ????? ????????? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/03abfb57/attachment.html From gozdal at gmail.com Wed Jan 23 12:48:14 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Wed, 23 Jan 2013 10:48:14 +0100 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: <50FE1857.7020804@communicatefreely.net> Message-ID: I'd doubt there is any VoIP phone with such support. But I'd guess writing tests querying actual phones wouldn't be much more work than: a) testing what kind of messages for BLF/MWI actually work on given phone b) testing (in ESL?) if FS sends good messages when certain actions are done (leaving voicemail or making a call) 2013/1/22 Michael Collins > That begs the question: does anyone know of any SIP phones that have an > API where you can poll the status of the lamps as well as what's on the > display, whether MWI is on, etc.? That would be epic. > > -MC > > On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre < > fs-list at communicatefreely.net> wrote: > >> Hmm. Good idea. >> >> I know that you can do this with Aastra phones, and possibly others. >> >> With the Aastra's, there is an XML API that you can use to push >> notifications, apps, etc. >> You can push XML to the phone on an HTTP interface (there is an ACL to >> tell it where to >> allow it from), and you can initiate phone calls, as well as hang up, >> transfer, etc. >> There is also an action URI, so you can have the phone report what is >> coming in on caller >> ID, when it is ringing, hung up, etc. >> >> I don't know if you can get lamp status from this or not, but perhaps you >> could have a >> system where a pattern of calls is generated, and after many repetitions, >> the system calls >> or a human to check and see that the lights match an expected state. >> >> Just an idea anyway. >> >> -Tim >> >> Michael Collins wrote: >> > This test suite does not presently exist. I would love to see some work >> > done in this area. >> > >> > One challenge is that there are numerous kinds of devices that can >> > interop with FS so the test suite will need to have some way to handle >> > the various phone makes and models. Also, I don't know which hard >> > phones, if any, support automated testing. I would *love* to see some >> > sort of web/network API on various phones where you could poll the phone >> > to see the state of each SLA/BLF key as well as what's in the display. >> > That would make automated testing much cleaner. As it stands now, >> > testing BLFs is a tedious process of watching phones and waiting to see >> > if the lights do what they should be doing. >> > >> > If anyone has practical suggestions (nothing pie-in-the-sky, if you get >> > my meaning) then by all means share with the group. >> > >> > -MC >> > >> > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine >> > > wrote: >> > >> > Hello, >> > >> > We are running version 1.0.6; we did not upgrade for a long time >> > due to some problems (mainly stuck BLFs) with the newer versions >> > which have been corrected a few months ago. >> > >> > A few weeks ago I pulled the latest GIT and did functional testings >> > on our development system; all was OK. >> > >> > Yesterday I moved our production system to that version. After a few >> > hours the problems started: stuck BLFs, hangups, LUA scripts >> > sometimes do not run. I had to revert back to the old version. >> > >> > My question: Is there some simple test suite which I can test >> > various scnarios on my development system? like, "call XX, while >> > this call is active call YY which is monitored by XX and hangup", >> > etc. It is hard to cover all cases manually, so I would like >> > something automated. >> > >> > Thanks, __Yehavi: >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Michael S Collins >> > Twitter: @mercutioviz >> > http://www.FreeSWITCH.org >> > http://www.ClueCon.com >> > http://www.OSTAG.org >> > >> > >> > ------------------------------------------------------------------------ >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Marcin Gozdalik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/4e5cb33b/attachment-0001.html From rostotskiy.a at gmail.com Wed Jan 23 13:46:55 2013 From: rostotskiy.a at gmail.com (Anton Rostotskiy) Date: Wed, 23 Jan 2013 12:46:55 +0200 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> Message-ID: Search results: /usr/local/freeswitch/opal/include/opal/localep.h /usr/local/src/freeswitch/libs/opal/include/ep/localep.h /usr/local/include/ep/localep.h /usr/include/ep/localep.h /root/opal/include/ep/localep.h As I said earlier, I installed ptlib and opal in /root directory. 2013/1/23 Steven Ayre > It's localep.h, not local.h: > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.h:41:26: > error: opal/localep.h: No such file or directory > > find / -name localep.h > > Sorry, seems my phone did an annoying autouncorrection! > > -Steve > > > > On 23 January 2013 08:16, Anton Rostotskiy wrote: > >> Thanks for reply, Steven! >> Tried to search local.h file, but search returned nothing. That are all >> local.h files in my system: >> >> /usr/local/src/freeswitch/libs/spandsp/src/gsm0610_local.h >> /usr/local/src/freeswitch/libs/spandsp/src/t30_local.h >> /usr/local/src/freeswitch/libs/spandsp/src/t42_t43_local.h >> >> About -devel packages - at first tried to install ptlib, opal and -devel >> from yum but their versions have been too low for mod_opal. >> >> 2013/1/22 Steven Ayre >> >>> Run: >>> find / -name local.h >>> >>> It's failing to include opal/local.h. All the other errors are because >>> its missing the definitions in that file. >>> >>> Normally I would say to instal the -devel package, but since you built >>> from svn that doesn't apply. Perhaps it's not installed or installed to the >>> wrong place. The find command will tell us where that file is. >>> >>> Steve on iPhone >>> >>> On 22 Jan 2013, at 16:07, Anton Rostotskiy >>> wrote: >>> >>> Also tried to install mod_opal and also failed. >>> CentOS 6.3 x86_64 >>> ptlib and opal are the last version obtained from svn as it is said in >>> wiki . >>> but just a bunch of errors after make mod_opal >>> http://pastebin.com/HHUPLCLV >>> >>> 2013/1/18 Steven Ayre >>> >>>> The latest version of opal in Debian (even in Sid) isn't new enough for >>>> mod_opal. It needed various features added that are only only available in >>>> the svn version. >>>> >>>> The wiki page states you have to install ptlib and opal from svn - this >>>> is why. >>>> http://wiki.freeswitch.org/wiki/Mod_opal >>>> >>>> Of course that can then cause conflicts unless you package it yourself >>>> or install a dummy package, or problems with other libraries/apps that are >>>> built against the libopal that's already packaged. >>>> >>>> Ptlib and h323plus/opal are also notorious for having to get matching >>>> versions. If you don't pair exactly the right version of ptlib with the >>>> specific version of h323plus/opal you can get instability problems. So the >>>> official Debian ptlib package probably wouldn't be good with opal from >>>> svn... >>>> >>>> As for mod_h323, it does compile and work, at least on Wheezy and >>>> possibly Squeeze too. I've never had any luck getting mod_opal working but >>>> have got mod_h323 working. I don't actively use it though - all my traffic >>>> is currently SIP. >>>> >>>> Another option is to run Yate as a SIP->H323 proxy. It only handles the >>>> signalling, media still goes directly caller->FS. It worked quite well, >>>> although it would occasionally crash - but at least that only dropped the >>>> H323 calls and didn't take FS down with it, and Yate automatically >>>> restarted. That was on Lenny so a long time ago... >>>> >>>> Steve on iPhone >>>> >>>> >>>> >>>> On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote: >>>> >>>> Hi, >>>> >>>> I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and >>>> opal >>>> 3.10.7 on a debian >>>> Ptlib and opal get build with no error, but when trying mod_opal I get >>>> >>>> Compiling >>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... >>>> quiet_libtool: compile: g++ -g -ggdb >>>> -I/usr/local/src/freeswitch/libs/curl/include >>>> -I/usr/local/src/freeswitch/src/include >>>> -I/usr/local/src/freeswitch/src/include >>>> -I/usr/local/src/freeswitch/libs/libteletone/src >>>> -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden >>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE >>>> -DP_64BIT >>>> -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions >>>> -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL >>>> -DHAVE_CONFIG_H -c >>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC >>>> -DPIC -o .libs/mod_opal.o >>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In >>>> member >>>> function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, >>>> switch_frame_t**, switch_io_flag_t)': >>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: >>>> error: 'class OpalCall' has no member named 'IsSwitchingT38' >>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In >>>> member >>>> function 'switch_status_t FSConnection::write_frame(const >>>> OpalMediaType&, >>>> const switch_frame_t*, switch_io_flag_t)': >>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: >>>> error: 'class OpalCall' has no member named 'IsSwitchingT38' >>>> make[3]: *** [mod_opal.lo] Error 1 >>>> make[2]: *** [install] Error 1 >>>> make[1]: *** [mod_opal-install] Error 1 >>>> make: *** [mod_opal-install] Error 2 >>>> >>>> >>>> Has anyone succeded in building mod_opal ? >>>> >>>> Should I go with mod_h323 instead ? >>>> >>>> Thanks for your help >>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http:// >>>> lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> ? ?????????, ????? ????????? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> ? ?????????, ????? ????????? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ? ?????????, ????? ????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/bc47aa0b/attachment-0001.html From steveayre at gmail.com Wed Jan 23 14:25:58 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 23 Jan 2013 11:25:58 +0000 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> Message-ID: Do you mean you built opal in /root/opal? You shouldn't build things as root, but that's not the reason for your troubles. I have to say I'm a bit puzzled, it looks like you've installed opal to multiple locations. /usr/local/freeswitch/opal looks the best match (opal/localep.h) but it's in the wrong location. /usr/local/src/freeswitch/libs/opal/include/ep/localep.h Not sure why it's installed here, opal isn't shipped under libs/ /usr/local/include/ep/localep.h /usr/include/ep/localep.h Close but the directory is wrong as it needs to be opal/localep.h, not ep/localep.h /usr/include/opal/localep.h Should be the correct filename, but you don't have a copy. It could be a change in the opal library but I find that unlikely since it'd break apps using the library like FS. More likely there's a problem with the prefixes given to opal's configure command. -Steve On 23 January 2013 10:46, Anton Rostotskiy wrote: > Search results: > > /usr/local/freeswitch/opal/include/opal/localep.h > /usr/local/src/freeswitch/libs/opal/include/ep/localep.h > /usr/local/include/ep/localep.h > /usr/include/ep/localep.h > /root/opal/include/ep/localep.h > > As I said earlier, I installed ptlib and opal in /root directory. > > 2013/1/23 Steven Ayre > >> It's localep.h, not local.h: >> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.h:41:26: >> error: opal/localep.h: No such file or directory >> >> find / -name localep.h >> >> Sorry, seems my phone did an annoying autouncorrection! >> >> -Steve >> >> >> >> On 23 January 2013 08:16, Anton Rostotskiy wrote: >> >>> Thanks for reply, Steven! >>> Tried to search local.h file, but search returned nothing. That are all >>> local.h files in my system: >>> >>> /usr/local/src/freeswitch/libs/spandsp/src/gsm0610_local.h >>> /usr/local/src/freeswitch/libs/spandsp/src/t30_local.h >>> /usr/local/src/freeswitch/libs/spandsp/src/t42_t43_local.h >>> >>> About -devel packages - at first tried to install ptlib, opal and -devel >>> from yum but their versions have been too low for mod_opal. >>> >>> 2013/1/22 Steven Ayre >>> >>>> Run: >>>> find / -name local.h >>>> >>>> It's failing to include opal/local.h. All the other errors are because >>>> its missing the definitions in that file. >>>> >>>> Normally I would say to instal the -devel package, but since you built >>>> from svn that doesn't apply. Perhaps it's not installed or installed to the >>>> wrong place. The find command will tell us where that file is. >>>> >>>> Steve on iPhone >>>> >>>> On 22 Jan 2013, at 16:07, Anton Rostotskiy >>>> wrote: >>>> >>>> Also tried to install mod_opal and also failed. >>>> CentOS 6.3 x86_64 >>>> ptlib and opal are the last version obtained from svn as it is said >>>> in wiki . >>>> but just a bunch of errors after make mod_opal >>>> http://pastebin.com/HHUPLCLV >>>> >>>> 2013/1/18 Steven Ayre >>>> >>>>> The latest version of opal in Debian (even in Sid) isn't new enough >>>>> for mod_opal. It needed various features added that are only only available >>>>> in the svn version. >>>>> >>>>> The wiki page states you have to install ptlib and opal from svn - >>>>> this is why. >>>>> http://wiki.freeswitch.org/wiki/Mod_opal >>>>> >>>>> Of course that can then cause conflicts unless you package it yourself >>>>> or install a dummy package, or problems with other libraries/apps that are >>>>> built against the libopal that's already packaged. >>>>> >>>>> Ptlib and h323plus/opal are also notorious for having to get matching >>>>> versions. If you don't pair exactly the right version of ptlib with the >>>>> specific version of h323plus/opal you can get instability problems. So the >>>>> official Debian ptlib package probably wouldn't be good with opal from >>>>> svn... >>>>> >>>>> As for mod_h323, it does compile and work, at least on Wheezy and >>>>> possibly Squeeze too. I've never had any luck getting mod_opal working but >>>>> have got mod_h323 working. I don't actively use it though - all my traffic >>>>> is currently SIP. >>>>> >>>>> Another option is to run Yate as a SIP->H323 proxy. It only handles >>>>> the signalling, media still goes directly caller->FS. It worked quite well, >>>>> although it would occasionally crash - but at least that only dropped the >>>>> H323 calls and didn't take FS down with it, and Yate automatically >>>>> restarted. That was on Lenny so a long time ago... >>>>> >>>>> Steve on iPhone >>>>> >>>>> >>>>> >>>>> On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote: >>>>> >>>>> Hi, >>>>> >>>>> I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and >>>>> opal >>>>> 3.10.7 on a debian >>>>> Ptlib and opal get build with no error, but when trying mod_opal I get >>>>> >>>>> Compiling >>>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... >>>>> quiet_libtool: compile: g++ -g -ggdb >>>>> -I/usr/local/src/freeswitch/libs/curl/include >>>>> -I/usr/local/src/freeswitch/src/include >>>>> -I/usr/local/src/freeswitch/src/include >>>>> -I/usr/local/src/freeswitch/libs/libteletone/src >>>>> -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden >>>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE >>>>> -DP_64BIT >>>>> -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions >>>>> -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL >>>>> -DHAVE_CONFIG_H -c >>>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp >>>>> -fPIC >>>>> -DPIC -o .libs/mod_opal.o >>>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In >>>>> member >>>>> function 'switch_status_t FSConnection::read_frame(const >>>>> OpalMediaType&, >>>>> switch_frame_t**, switch_io_flag_t)': >>>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: >>>>> error: 'class OpalCall' has no member named 'IsSwitchingT38' >>>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In >>>>> member >>>>> function 'switch_status_t FSConnection::write_frame(const >>>>> OpalMediaType&, >>>>> const switch_frame_t*, switch_io_flag_t)': >>>>> /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: >>>>> error: 'class OpalCall' has no member named 'IsSwitchingT38' >>>>> make[3]: *** [mod_opal.lo] Error 1 >>>>> make[2]: *** [install] Error 1 >>>>> make[1]: *** [mod_opal-install] Error 1 >>>>> make: *** [mod_opal-install] Error 2 >>>>> >>>>> >>>>> Has anyone succeded in building mod_opal ? >>>>> >>>>> Should I go with mod_h323 instead ? >>>>> >>>>> Thanks for your help >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http:// >>>>> lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> ? ?????????, ????? ????????? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http:// >>>> lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> ? ?????????, ????? ????????? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > ? ?????????, ????? ????????? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/7b917d02/attachment-0001.html From regis.freeswitch.org at tornad.net Wed Jan 23 15:18:43 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Wed, 23 Jan 2013 13:18:43 +0100 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: <50FE1857.7020804@communicatefreely.net> Message-ID: Maybe It's could be done with some sip session record that we can playback to freeswitch ? In this mode, sofia will ignore or bypass time announce and sequence number to anwer the correct/same thing as the first "real" recording sequence. A sort of mod_sofia_test, one FS can read pcap or sip sequence and throw mod_sofia_test "simulate" the endpoint to another or itself. It could allow to record all phone session (register, makecall, holdcall, transfer,...) and check FS still answer same things or corrects thing. A little hard maybe ? :) I'm not a specialist of "sequence" or SIP change that occurs in a session dialog between 2 calls from exactly the same device. 2013/1/23 Marcin Gozdalik > I'd doubt there is any VoIP phone with such support. > > But I'd guess writing tests querying actual phones wouldn't be much more > work than: > a) testing what kind of messages for BLF/MWI actually work on given phone > b) testing (in ESL?) if FS sends good messages when certain actions are > done (leaving voicemail or making a call) > > > > > 2013/1/22 Michael Collins > >> That begs the question: does anyone know of any SIP phones that have an >> API where you can poll the status of the lamps as well as what's on the >> display, whether MWI is on, etc.? That would be epic. >> >> -MC >> >> On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre < >> fs-list at communicatefreely.net> wrote: >> >>> Hmm. Good idea. >>> >>> I know that you can do this with Aastra phones, and possibly others. >>> >>> With the Aastra's, there is an XML API that you can use to push >>> notifications, apps, etc. >>> You can push XML to the phone on an HTTP interface (there is an ACL to >>> tell it where to >>> allow it from), and you can initiate phone calls, as well as hang up, >>> transfer, etc. >>> There is also an action URI, so you can have the phone report what is >>> coming in on caller >>> ID, when it is ringing, hung up, etc. >>> >>> I don't know if you can get lamp status from this or not, but perhaps >>> you could have a >>> system where a pattern of calls is generated, and after many >>> repetitions, the system calls >>> or a human to check and see that the lights match an expected state. >>> >>> Just an idea anyway. >>> >>> -Tim >>> >>> Michael Collins wrote: >>> > This test suite does not presently exist. I would love to see some work >>> > done in this area. >>> > >>> > One challenge is that there are numerous kinds of devices that can >>> > interop with FS so the test suite will need to have some way to handle >>> > the various phone makes and models. Also, I don't know which hard >>> > phones, if any, support automated testing. I would *love* to see some >>> > sort of web/network API on various phones where you could poll the >>> phone >>> > to see the state of each SLA/BLF key as well as what's in the display. >>> > That would make automated testing much cleaner. As it stands now, >>> > testing BLFs is a tedious process of watching phones and waiting to see >>> > if the lights do what they should be doing. >>> > >>> > If anyone has practical suggestions (nothing pie-in-the-sky, if you get >>> > my meaning) then by all means share with the group. >>> > >>> > -MC >>> > >>> > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine >>> > > wrote: >>> > >>> > Hello, >>> > >>> > We are running version 1.0.6; we did not upgrade for a long time >>> > due to some problems (mainly stuck BLFs) with the newer versions >>> > which have been corrected a few months ago. >>> > >>> > A few weeks ago I pulled the latest GIT and did functional testings >>> > on our development system; all was OK. >>> > >>> > Yesterday I moved our production system to that version. After a >>> few >>> > hours the problems started: stuck BLFs, hangups, LUA scripts >>> > sometimes do not run. I had to revert back to the old version. >>> > >>> > My question: Is there some simple test suite which I can test >>> > various scnarios on my development system? like, "call XX, while >>> > this call is active call YY which is monitored by XX and hangup", >>> > etc. It is hard to cover all cases manually, so I would like >>> > something automated. >>> > >>> > Thanks, __Yehavi: >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> > -- >>> > Michael S Collins >>> > Twitter: @mercutioviz >>> > http://www.FreeSWITCH.org >>> > http://www.ClueCon.com >>> > http://www.OSTAG.org >>> > >>> > >>> > >>> ------------------------------------------------------------------------ >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Marcin Gozdalik > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/212cef78/attachment.html From jmesquita at freeswitch.org Wed Jan 23 16:52:32 2013 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 23 Jan 2013 10:52:32 -0300 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: <50FE1857.7020804@communicatefreely.net> Message-ID: That is exactly what sipp does but setting up the scenarios requires some knowledge but basically it will assemble the packets as you want on scenarios and send them out to FreeSWITCH. If it fails to receive an expected response, it will complain. In my opinion, testing things with real devices is not feasible, specially if you are looking for automated testing that can be executes frequently. We are working on a simple idea that can grow to something bigger. Lets see how it goes. Sent from my iPhone On Jan 23, 2013, at 9:18 AM, Regis M wrote: > Maybe It's could be done with some sip session record that we can playback to freeswitch ? > > In this mode, sofia will ignore or bypass time announce and sequence number to anwer the correct/same thing as the first "real" recording sequence. > > A sort of mod_sofia_test, one FS can read pcap or sip sequence and throw mod_sofia_test "simulate" the endpoint to another or itself. > It could allow to record all phone session (register, makecall, holdcall, transfer,...) and check FS still answer same things or corrects thing. > > A little hard maybe ? :) > I'm not a specialist of "sequence" or SIP change that occurs in a session dialog between 2 calls from exactly the same device. > > > > > > 2013/1/23 Marcin Gozdalik >> I'd doubt there is any VoIP phone with such support. >> >> But I'd guess writing tests querying actual phones wouldn't be much more work than: >> a) testing what kind of messages for BLF/MWI actually work on given phone >> b) testing (in ESL?) if FS sends good messages when certain actions are done (leaving voicemail or making a call) >> >> >> >> >> 2013/1/22 Michael Collins >>> That begs the question: does anyone know of any SIP phones that have an API where you can poll the status of the lamps as well as what's on the display, whether MWI is on, etc.? That would be epic. >>> >>> -MC >>> >>> On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre wrote: >>>> Hmm. Good idea. >>>> >>>> I know that you can do this with Aastra phones, and possibly others. >>>> >>>> With the Aastra's, there is an XML API that you can use to push notifications, apps, etc. >>>> You can push XML to the phone on an HTTP interface (there is an ACL to tell it where to >>>> allow it from), and you can initiate phone calls, as well as hang up, transfer, etc. >>>> There is also an action URI, so you can have the phone report what is coming in on caller >>>> ID, when it is ringing, hung up, etc. >>>> >>>> I don't know if you can get lamp status from this or not, but perhaps you could have a >>>> system where a pattern of calls is generated, and after many repetitions, the system calls >>>> or a human to check and see that the lights match an expected state. >>>> >>>> Just an idea anyway. >>>> >>>> -Tim >>>> >>>> Michael Collins wrote: >>>> > This test suite does not presently exist. I would love to see some work >>>> > done in this area. >>>> > >>>> > One challenge is that there are numerous kinds of devices that can >>>> > interop with FS so the test suite will need to have some way to handle >>>> > the various phone makes and models. Also, I don't know which hard >>>> > phones, if any, support automated testing. I would *love* to see some >>>> > sort of web/network API on various phones where you could poll the phone >>>> > to see the state of each SLA/BLF key as well as what's in the display. >>>> > That would make automated testing much cleaner. As it stands now, >>>> > testing BLFs is a tedious process of watching phones and waiting to see >>>> > if the lights do what they should be doing. >>>> > >>>> > If anyone has practical suggestions (nothing pie-in-the-sky, if you get >>>> > my meaning) then by all means share with the group. >>>> > >>>> > -MC >>>> > >>>> > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine >>>> > > wrote: >>>> > >>>> > Hello, >>>> > >>>> > We are running version 1.0.6; we did not upgrade for a long time >>>> > due to some problems (mainly stuck BLFs) with the newer versions >>>> > which have been corrected a few months ago. >>>> > >>>> > A few weeks ago I pulled the latest GIT and did functional testings >>>> > on our development system; all was OK. >>>> > >>>> > Yesterday I moved our production system to that version. After a few >>>> > hours the problems started: stuck BLFs, hangups, LUA scripts >>>> > sometimes do not run. I had to revert back to the old version. >>>> > >>>> > My question: Is there some simple test suite which I can test >>>> > various scnarios on my development system? like, "call XX, while >>>> > this call is active call YY which is monitored by XX and hangup", >>>> > etc. It is hard to cover all cases manually, so I would like >>>> > something automated. >>>> > >>>> > Thanks, __Yehavi: >>>> > >>>> > _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> > >>>> > >>>> > -- >>>> > Michael S Collins >>>> > Twitter: @mercutioviz >>>> > http://www.FreeSWITCH.org >>>> > http://www.ClueCon.com >>>> > http://www.OSTAG.org >>>> > >>>> > >>>> > ------------------------------------------------------------------------ >>>> > >>>> > _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Marcin Gozdalik >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/1cb400ba/attachment-0001.html From bdfoster at endigotech.com Wed Jan 23 17:32:01 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 23 Jan 2013 09:32:01 -0500 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: <50FE1857.7020804@communicatefreely.net> Message-ID: Polycom has some stuff but probably not everything is exposed Sent from my iPhone On Jan 22, 2013, at 5:24 PM, Michael Collins wrote: > That begs the question: does anyone know of any SIP phones that have an API where you can poll the status of the lamps as well as what's on the display, whether MWI is on, etc.? That would be epic. > > -MC > > On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre wrote: >> Hmm. Good idea. >> >> I know that you can do this with Aastra phones, and possibly others. >> >> With the Aastra's, there is an XML API that you can use to push notifications, apps, etc. >> You can push XML to the phone on an HTTP interface (there is an ACL to tell it where to >> allow it from), and you can initiate phone calls, as well as hang up, transfer, etc. >> There is also an action URI, so you can have the phone report what is coming in on caller >> ID, when it is ringing, hung up, etc. >> >> I don't know if you can get lamp status from this or not, but perhaps you could have a >> system where a pattern of calls is generated, and after many repetitions, the system calls >> or a human to check and see that the lights match an expected state. >> >> Just an idea anyway. >> >> -Tim >> >> Michael Collins wrote: >> > This test suite does not presently exist. I would love to see some work >> > done in this area. >> > >> > One challenge is that there are numerous kinds of devices that can >> > interop with FS so the test suite will need to have some way to handle >> > the various phone makes and models. Also, I don't know which hard >> > phones, if any, support automated testing. I would *love* to see some >> > sort of web/network API on various phones where you could poll the phone >> > to see the state of each SLA/BLF key as well as what's in the display. >> > That would make automated testing much cleaner. As it stands now, >> > testing BLFs is a tedious process of watching phones and waiting to see >> > if the lights do what they should be doing. >> > >> > If anyone has practical suggestions (nothing pie-in-the-sky, if you get >> > my meaning) then by all means share with the group. >> > >> > -MC >> > >> > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine >> > > wrote: >> > >> > Hello, >> > >> > We are running version 1.0.6; we did not upgrade for a long time >> > due to some problems (mainly stuck BLFs) with the newer versions >> > which have been corrected a few months ago. >> > >> > A few weeks ago I pulled the latest GIT and did functional testings >> > on our development system; all was OK. >> > >> > Yesterday I moved our production system to that version. After a few >> > hours the problems started: stuck BLFs, hangups, LUA scripts >> > sometimes do not run. I had to revert back to the old version. >> > >> > My question: Is there some simple test suite which I can test >> > various scnarios on my development system? like, "call XX, while >> > this call is active call YY which is monitored by XX and hangup", >> > etc. It is hard to cover all cases manually, so I would like >> > something automated. >> > >> > Thanks, __Yehavi: >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Michael S Collins >> > Twitter: @mercutioviz >> > http://www.FreeSWITCH.org >> > http://www.ClueCon.com >> > http://www.OSTAG.org >> > >> > >> > ------------------------------------------------------------------------ >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/c1744ff4/attachment.html From bdfoster at endigotech.com Wed Jan 23 17:59:11 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 23 Jan 2013 09:59:11 -0500 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: <50FE1857.7020804@communicatefreely.net> Message-ID: I see further down the thread there's a mention of Aastra's polling feature as well. Polycom's implementation is similar, and I'd have to read the documentation but there's a chance you can get the status of a particular line. That might be what you're after. I would actually be interested in helping in this area, if we find out those polling features would be beneficial in troubleshooting, even if it doesn't apply to this particular case. Sent from my iPhone On Jan 22, 2013, at 5:24 PM, Michael Collins wrote: > That begs the question: does anyone know of any SIP phones that have an API where you can poll the status of the lamps as well as what's on the display, whether MWI is on, etc.? That would be epic. > > -MC > > On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre wrote: >> Hmm. Good idea. >> >> I know that you can do this with Aastra phones, and possibly others. >> >> With the Aastra's, there is an XML API that you can use to push notifications, apps, etc. >> You can push XML to the phone on an HTTP interface (there is an ACL to tell it where to >> allow it from), and you can initiate phone calls, as well as hang up, transfer, etc. >> There is also an action URI, so you can have the phone report what is coming in on caller >> ID, when it is ringing, hung up, etc. >> >> I don't know if you can get lamp status from this or not, but perhaps you could have a >> system where a pattern of calls is generated, and after many repetitions, the system calls >> or a human to check and see that the lights match an expected state. >> >> Just an idea anyway. >> >> -Tim >> >> Michael Collins wrote: >> > This test suite does not presently exist. I would love to see some work >> > done in this area. >> > >> > One challenge is that there are numerous kinds of devices that can >> > interop with FS so the test suite will need to have some way to handle >> > the various phone makes and models. Also, I don't know which hard >> > phones, if any, support automated testing. I would *love* to see some >> > sort of web/network API on various phones where you could poll the phone >> > to see the state of each SLA/BLF key as well as what's in the display. >> > That would make automated testing much cleaner. As it stands now, >> > testing BLFs is a tedious process of watching phones and waiting to see >> > if the lights do what they should be doing. >> > >> > If anyone has practical suggestions (nothing pie-in-the-sky, if you get >> > my meaning) then by all means share with the group. >> > >> > -MC >> > >> > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine >> > > wrote: >> > >> > Hello, >> > >> > We are running version 1.0.6; we did not upgrade for a long time >> > due to some problems (mainly stuck BLFs) with the newer versions >> > which have been corrected a few months ago. >> > >> > A few weeks ago I pulled the latest GIT and did functional testings >> > on our development system; all was OK. >> > >> > Yesterday I moved our production system to that version. After a few >> > hours the problems started: stuck BLFs, hangups, LUA scripts >> > sometimes do not run. I had to revert back to the old version. >> > >> > My question: Is there some simple test suite which I can test >> > various scnarios on my development system? like, "call XX, while >> > this call is active call YY which is monitored by XX and hangup", >> > etc. It is hard to cover all cases manually, so I would like >> > something automated. >> > >> > Thanks, __Yehavi: >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Michael S Collins >> > Twitter: @mercutioviz >> > http://www.FreeSWITCH.org >> > http://www.ClueCon.com >> > http://www.OSTAG.org >> > >> > >> > ------------------------------------------------------------------------ >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/e143b60d/attachment-0001.html From danb.lists at gmail.com Wed Jan 23 20:07:13 2013 From: danb.lists at gmail.com (DanB) Date: Wed, 23 Jan 2013 18:07:13 +0100 Subject: [Freeswitch-users] mod_gsmopen with Motorola i465 Message-ID: <510018C1.9080102@gmail.com> Hey Guys (especially Giovanni), I am trying to use mod_gsmopen - the mobigater flavor - with a Motorola i465. Before opening a jira I was wondering if anyone had any experience with this phone and if there are chances to make it work? I can see some information being exchanged when I try to dial out via originate command but it fails at some point after sending "AT+MMGL="HEADER ONLY" (see bellow). I should mention that the phone is CDMA version. Further logs/debug available on request. """ 2013-01-23 11:19:50.193765 [DEBUG] gsmopen_protocol.c:2360 rev [(nil)|37 ][DEBUG_GSMOPEN 2360 ][interface1][-1,24, 3] sending: AT+MMGL="HEADER ONLY" 2013-01-23 11:19:50.233538 [DEBUG] gsmopen_protocol.c:1078 rev [(nil)|37 ][DEBUG_GSMOPEN 1078 ][interface1][-1,24, 3] Read line 0: |ERROR| 2013-01-23 11:19:50.233538 [DEBUG] gsmopen_protocol.c:1943 rev [(nil)|37 ][DEBUG_GSMOPEN 1943 ][interface1][-1,24, 3] got ERROR 2013-01-23 11:19:50.233538 [WARNING] gsmopen_protocol.c:3911 rev [(nil)|37 ][WARNINGA 3911 ][interface1][-1,24, 3] AT+MMGL="HEADER ONLY" does not get OK from the phone. If your phone is not Motorola, please contact the gsmopen developers. Else, if your phone IS a Motorola, probably a long msg was incoming and ther first part was read and then deleted. The second part is now orphan. If you got this warning repeatedly, and you cannot correctly receive SMSs from this interface, please manually clean all messages (and the residual parts of them) from the cellphone/SIM. Continuing. """ Thanks in advance for any kind of tip! DanB From msc at freeswitch.org Wed Jan 23 20:16:04 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Jan 2013 09:16:04 -0800 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today Message-ID: Hello folks, On today's call we have a few news items to discuss, after which we will have an open forum for Q&A on FreeSWITCH topics. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_23 Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/93c37bf1/attachment.html From msc at freeswitch.org Wed Jan 23 20:19:55 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Jan 2013 09:19:55 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> Message-ID: I find it interesting that the ZRTP engine threw a warning. Do you see any ZRTP messages on the Linux log where the fax actually worked? -MC On Tue, Jan 22, 2013 at 2:35 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > Here's an update on my FAX project: > > On our main FreeSWITCH pbx (on CentOS), I run this command through the CLI > and it works: > > freeswitch at internal> originate sofia/gateway/flowroute/1408739xxxx > &txfax(/home/schoch/fax000007603.tif) > > I did a basic install on a Windows machine, by downloading the installer, > installing it, and changing the "default_provider" lines in vars.xml in > order to make it connect to our pbx. A "sofia status" shows it's connected. > > Then I ran this command on the Windows machine: > freeswitch at internal> originate > sofia/gateway/external::starnet_fax/1408739xxxx > &txfax(C:/Users/Schoch/Pictures/fax000007603.tif) > > It rang the FAX machine, but the FAX never got there. It got some sort of > a timeout: > 2013-01-22 14:09:10.224456 [DEBUG] switch_rtp.c:928 [ zrtp engine]: > WARNING! HELLO Max retransmissions count reached (20 retries). ID=1 > and > 2013-01-22 14:10:07.744746 [DEBUG] mod_spandsp_fax.c:500 Fax processing > not successful - result (2) Timed out waiting for initial communication. > > I'm not sure where to go from here. Any ideas? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/1b7ae66d/attachment.html From gmaruzz at gmail.com Wed Jan 23 20:21:36 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 23 Jan 2013 18:21:36 +0100 Subject: [Freeswitch-users] mod_gsmopen with Motorola i465 In-Reply-To: <510018C1.9080102@gmail.com> References: <510018C1.9080102@gmail.com> Message-ID: On Wed, Jan 23, 2013 at 6:07 PM, DanB wrote: > Hey Guys (especially Giovanni), > > I am trying to use mod_gsmopen - the mobigater flavor - with a Motorola > i465. I would suggest you to forget about :). Use those cheap Huawei and be happy, or use whatever else and be prepared to enter a world of pain, and to devote an incredible amount of time to it. (been there done that) -giovanni > > Before opening a jira I was wondering if anyone had any experience with > this phone and if there are chances to make it work? I can see some > information being exchanged when I try to dial out via originate command > but it fails at some point after sending "AT+MMGL="HEADER ONLY" (see > bellow). I should mention that the phone is CDMA version. Further > logs/debug available on request. > > """ > 2013-01-23 11:19:50.193765 [DEBUG] gsmopen_protocol.c:2360 rev [(nil)|37 > ][DEBUG_GSMOPEN 2360 ][interface1][-1,24, 3] sending: > AT+MMGL="HEADER ONLY" > 2013-01-23 11:19:50.233538 [DEBUG] gsmopen_protocol.c:1078 rev [(nil)|37 > ][DEBUG_GSMOPEN 1078 ][interface1][-1,24, 3] Read line 0: |ERROR| > 2013-01-23 11:19:50.233538 [DEBUG] gsmopen_protocol.c:1943 rev [(nil)|37 > ][DEBUG_GSMOPEN 1943 ][interface1][-1,24, 3] got ERROR > 2013-01-23 11:19:50.233538 [WARNING] gsmopen_protocol.c:3911 rev > [(nil)|37 ][WARNINGA 3911 ][interface1][-1,24, 3] AT+MMGL="HEADER > ONLY" does not get OK from the phone. If your phone is not Motorola, > please contact the gsmopen developers. Else, if your phone IS a > Motorola, probably a long msg was incoming and ther first part was read > and then deleted. The second part is now orphan. If you got this warning > repeatedly, and you cannot correctly receive SMSs from this interface, > please manually clean all messages (and the residual parts of them) from > the cellphone/SIM. Continuing. > """ > > Thanks in advance for any kind of tip! > > DanB > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From msc at freeswitch.org Wed Jan 23 20:25:06 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Jan 2013 09:25:06 -0800 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: <50FE1857.7020804@communicatefreely.net> Message-ID: After 4 years of working on the CudaTel team and hearing about SLA and BLF failures I can tell you that you need to see what's actually displayed on the phone itself, so checking the SIP traffic is not sufficient for a thorough test. If anyone is willing to crack open a phone and do some wiring they might be able to use a raspi or arduino and some components to monitor the phone. It's not the easiest thing in the world to do. That's why having a phone with an API would be so helpful. Autotesting would save enormous amounts of time in the long run. -MC On Wed, Jan 23, 2013 at 4:18 AM, Regis M wrote: > Maybe It's could be done with some sip session record that we can playback > to freeswitch ? > > In this mode, sofia will ignore or bypass time announce and sequence > number to anwer the correct/same thing as the first "real" recording > sequence. > > A sort of mod_sofia_test, one FS can read pcap or sip sequence and throw > mod_sofia_test "simulate" the endpoint to another or itself. > It could allow to record all phone session (register, makecall, holdcall, > transfer,...) and check FS still answer same things or corrects thing. > > A little hard maybe ? :) > I'm not a specialist of "sequence" or SIP change that occurs in a session > dialog between 2 calls from exactly the same device. > > > > > > 2013/1/23 Marcin Gozdalik > >> I'd doubt there is any VoIP phone with such support. >> >> But I'd guess writing tests querying actual phones wouldn't be much more >> work than: >> a) testing what kind of messages for BLF/MWI actually work on given phone >> b) testing (in ESL?) if FS sends good messages when certain actions are >> done (leaving voicemail or making a call) >> >> >> >> >> 2013/1/22 Michael Collins >> >>> That begs the question: does anyone know of any SIP phones that have an >>> API where you can poll the status of the lamps as well as what's on the >>> display, whether MWI is on, etc.? That would be epic. >>> >>> -MC >>> >>> On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre < >>> fs-list at communicatefreely.net> wrote: >>> >>>> Hmm. Good idea. >>>> >>>> I know that you can do this with Aastra phones, and possibly others. >>>> >>>> With the Aastra's, there is an XML API that you can use to push >>>> notifications, apps, etc. >>>> You can push XML to the phone on an HTTP interface (there is an ACL >>>> to tell it where to >>>> allow it from), and you can initiate phone calls, as well as hang up, >>>> transfer, etc. >>>> There is also an action URI, so you can have the phone report what is >>>> coming in on caller >>>> ID, when it is ringing, hung up, etc. >>>> >>>> I don't know if you can get lamp status from this or not, but perhaps >>>> you could have a >>>> system where a pattern of calls is generated, and after many >>>> repetitions, the system calls >>>> or a human to check and see that the lights match an expected state. >>>> >>>> Just an idea anyway. >>>> >>>> -Tim >>>> >>>> Michael Collins wrote: >>>> > This test suite does not presently exist. I would love to see some >>>> work >>>> > done in this area. >>>> > >>>> > One challenge is that there are numerous kinds of devices that can >>>> > interop with FS so the test suite will need to have some way to handle >>>> > the various phone makes and models. Also, I don't know which hard >>>> > phones, if any, support automated testing. I would *love* to see some >>>> > sort of web/network API on various phones where you could poll the >>>> phone >>>> > to see the state of each SLA/BLF key as well as what's in the display. >>>> > That would make automated testing much cleaner. As it stands now, >>>> > testing BLFs is a tedious process of watching phones and waiting to >>>> see >>>> > if the lights do what they should be doing. >>>> > >>>> > If anyone has practical suggestions (nothing pie-in-the-sky, if you >>>> get >>>> > my meaning) then by all means share with the group. >>>> > >>>> > -MC >>>> > >>>> > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine >>>> > > wrote: >>>> > >>>> > Hello, >>>> > >>>> > We are running version 1.0.6; we did not upgrade for a long time >>>> > due to some problems (mainly stuck BLFs) with the newer versions >>>> > which have been corrected a few months ago. >>>> > >>>> > A few weeks ago I pulled the latest GIT and did functional >>>> testings >>>> > on our development system; all was OK. >>>> > >>>> > Yesterday I moved our production system to that version. After a >>>> few >>>> > hours the problems started: stuck BLFs, hangups, LUA scripts >>>> > sometimes do not run. I had to revert back to the old version. >>>> > >>>> > My question: Is there some simple test suite which I can test >>>> > various scnarios on my development system? like, "call XX, while >>>> > this call is active call YY which is monitored by XX and hangup", >>>> > etc. It is hard to cover all cases manually, so I would like >>>> > something automated. >>>> > >>>> > Thanks, __Yehavi: >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> > >>>> > >>>> > -- >>>> > Michael S Collins >>>> > Twitter: @mercutioviz >>>> > http://www.FreeSWITCH.org >>>> > http://www.ClueCon.com >>>> > http://www.OSTAG.org >>>> > >>>> > >>>> > >>>> ------------------------------------------------------------------------ >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Marcin Gozdalik >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/6e6b4d0e/attachment-0001.html From danb.lists at gmail.com Wed Jan 23 20:34:58 2013 From: danb.lists at gmail.com (DanB) Date: Wed, 23 Jan 2013 18:34:58 +0100 Subject: [Freeswitch-users] mod_gsmopen with Motorola i465 In-Reply-To: References: Message-ID: <51001F42.6050201@gmail.com> Giovanni, Thank you for your so fast answer. Do I have any chance finding a CDMA phone working with your module then? DanB From gmaruzz at gmail.com Wed Jan 23 20:44:20 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 23 Jan 2013 18:44:20 +0100 Subject: [Freeswitch-users] mod_gsmopen with Motorola i465 In-Reply-To: <51001F42.6050201@gmail.com> References: <51001F42.6050201@gmail.com> Message-ID: unfortunately I don't know about. On 1/23/13, DanB wrote: > Giovanni, > > Thank you for your so fast answer. Do I have any chance finding a CDMA > phone working with your module then? > > DanB > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From raimund.sacherer at logitravel.com Wed Jan 23 20:47:33 2013 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Wed, 23 Jan 2013 18:47:33 +0100 (CET) Subject: [Freeswitch-users] mod_stress In-Reply-To: <18262686.1519.1358963234873.JavaMail.javamailuser@localhost> Message-ID: <304825.1523.1358963241421.JavaMail.javamailuser@localhost> Hi Peter, can you also make some statistics available on how much this module, if activated, adds to the ressource usage? If you have a busy callcenter, would this make a noticable impact on CPU usage? Also, would this use an additional G729 codec for calls using this codec? thanks, best regards Raimund ----- Original Message ----- From: "Peter Steinbach" To: "FreeSWITCH Users Help" Sent: Mi?rcoles, 23 de Enero 2013 0:34:41 Subject: Re: [Freeswitch-users] mod_stress Thanks Steven, I got it working and reported my howto and findings in the wiki: http://wiki.freeswitch.org/wiki/Mod_stress I found it to be a nice application, but as you explained, there are some false positives. Best regards Peter On 01/22/13 12:39, Steven Ayre wrote: Media bugs are used to hook into the call and provide decoded audio to the module. That requires transcoding, naturally. See stress_callback which is called for each audio frame plus some init/cleanup hooks, and read up on the API at http://docs.freeswitch.org/group__mb1.html . I suspect read vs write would correspond to the speaking/listening directions of the channel. Read from caller, write to caller? The names come from the media bug API. The read/write options will add the corresponding media bug, stop will remove them. SWITCH_ADD_APP tells you it's a dialplan application - it's started from dialplan, although you can also trigger dialplan apps on a channel via ESL using uuid_broadcast. The FFT algorithm is used to extract the frequencies present in the sample (the power spectrum). The algorithm then tries to detect stressed based on that. Looks like the algorithm at least in part attempts to look for tremor in the voice. More tremor indicates more stress. Over a threshold it raises an event. Bear in mind that as it's pattern recognition (and little tested at that) you'll get a lot of false positives and false negatives. It's also looking for something in the voice that may or may not be there - some people might be able to speak with an unstressed voice when they're actually quite stressed. -Steve On 22 January 2013 11:19, Peter Steinbach < lists at telefaks.de > wrote:
Hello Anthony, although it may be incomplete, it may be useful for us. Can you give us some basic explanation, how this works? I can see in the code that there is a stress command with read|write|stop parameter. Is this to be issued via event socket or is it a dialplan application? I suspect that "read" somehow starts the detection and "stop" somehow stops the detection. For "write" I currently have no idea. I would be happy to test this and report my configuration, explanations and examples in the wiki. Best regards Peter On 01/22/13 00:30, Anthony Minessale wrote:
That module is incomplete we made it several years ago to test some code that was supposed to pick up on stress in your voice in a media bug so we could fire events and whisper "lie" to the caller as he talked to the guy. On Mon, Jan 21, 2013 at 4:56 PM, Peter Steinbach < lists at telefaks.de > wrote:
I am also wondering, what this module does in detail. Regarding the source code, I can see that FFT's are made in order to to get some sort of frequency profile. But the code and remarks do not give me a clue how this module actually works. Best regards Peter On 05/26/12 15:05, BF wrote: > Hello, > > I'd like to know what the "stress write" from mod_stress module does. I have hard time finding documentation or an example about mod_stress module. > > Anyone? > > Thank you. > > Eric > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto: lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn: +19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/7cd6d138/attachment-0001.html From yehavi.bourvine at gmail.com Wed Jan 23 21:02:16 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 23 Jan 2013 20:02:16 +0200 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: <50FE1857.7020804@communicatefreely.net> Message-ID: I was looking for somehting simpler: Create a "script" that will do some calls in parallel in various sequences; it would be run while I watch the BLFs. When I see that a test has ended with stuck BLF and it is consistent, then I can strip down the test until I find where the problem is. This test has two benefits: testing before moving to production, and if fails - a way to repeat it. Thanks, __Yehavi: 2013/1/23 Michael Collins > After 4 years of working on the CudaTel team and hearing about SLA and BLF > failures I can tell you that you need to see what's actually displayed on > the phone itself, so checking the SIP traffic is not sufficient for a > thorough test. > > If anyone is willing to crack open a phone and do some wiring they might > be able to use a raspi or arduino and some components to monitor the phone. > It's not the easiest thing in the world to do. That's why having a phone > with an API would be so helpful. Autotesting would save enormous amounts of > time in the long run. > > -MC > > > On Wed, Jan 23, 2013 at 4:18 AM, Regis M wrote: > >> Maybe It's could be done with some sip session record that we can >> playback to freeswitch ? >> >> In this mode, sofia will ignore or bypass time announce and sequence >> number to anwer the correct/same thing as the first "real" recording >> sequence. >> >> A sort of mod_sofia_test, one FS can read pcap or sip sequence and throw >> mod_sofia_test "simulate" the endpoint to another or itself. >> It could allow to record all phone session (register, makecall, holdcall, >> transfer,...) and check FS still answer same things or corrects thing. >> >> A little hard maybe ? :) >> I'm not a specialist of "sequence" or SIP change that occurs in a session >> dialog between 2 calls from exactly the same device. >> >> >> >> >> >> 2013/1/23 Marcin Gozdalik >> >>> I'd doubt there is any VoIP phone with such support. >>> >>> But I'd guess writing tests querying actual phones wouldn't be much more >>> work than: >>> a) testing what kind of messages for BLF/MWI actually work on given phone >>> b) testing (in ESL?) if FS sends good messages when certain actions are >>> done (leaving voicemail or making a call) >>> >>> >>> >>> >>> 2013/1/22 Michael Collins >>> >>>> That begs the question: does anyone know of any SIP phones that have an >>>> API where you can poll the status of the lamps as well as what's on the >>>> display, whether MWI is on, etc.? That would be epic. >>>> >>>> -MC >>>> >>>> On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre < >>>> fs-list at communicatefreely.net> wrote: >>>> >>>>> Hmm. Good idea. >>>>> >>>>> I know that you can do this with Aastra phones, and possibly others. >>>>> >>>>> With the Aastra's, there is an XML API that you can use to push >>>>> notifications, apps, etc. >>>>> You can push XML to the phone on an HTTP interface (there is an ACL >>>>> to tell it where to >>>>> allow it from), and you can initiate phone calls, as well as hang up, >>>>> transfer, etc. >>>>> There is also an action URI, so you can have the phone report what is >>>>> coming in on caller >>>>> ID, when it is ringing, hung up, etc. >>>>> >>>>> I don't know if you can get lamp status from this or not, but perhaps >>>>> you could have a >>>>> system where a pattern of calls is generated, and after many >>>>> repetitions, the system calls >>>>> or a human to check and see that the lights match an expected state. >>>>> >>>>> Just an idea anyway. >>>>> >>>>> -Tim >>>>> >>>>> Michael Collins wrote: >>>>> > This test suite does not presently exist. I would love to see some >>>>> work >>>>> > done in this area. >>>>> > >>>>> > One challenge is that there are numerous kinds of devices that can >>>>> > interop with FS so the test suite will need to have some way to >>>>> handle >>>>> > the various phone makes and models. Also, I don't know which hard >>>>> > phones, if any, support automated testing. I would *love* to see some >>>>> > sort of web/network API on various phones where you could poll the >>>>> phone >>>>> > to see the state of each SLA/BLF key as well as what's in the >>>>> display. >>>>> > That would make automated testing much cleaner. As it stands now, >>>>> > testing BLFs is a tedious process of watching phones and waiting to >>>>> see >>>>> > if the lights do what they should be doing. >>>>> > >>>>> > If anyone has practical suggestions (nothing pie-in-the-sky, if you >>>>> get >>>>> > my meaning) then by all means share with the group. >>>>> > >>>>> > -MC >>>>> > >>>>> > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine >>>>> > > >>>>> wrote: >>>>> > >>>>> > Hello, >>>>> > >>>>> > We are running version 1.0.6; we did not upgrade for a long >>>>> time >>>>> > due to some problems (mainly stuck BLFs) with the newer versions >>>>> > which have been corrected a few months ago. >>>>> > >>>>> > A few weeks ago I pulled the latest GIT and did functional >>>>> testings >>>>> > on our development system; all was OK. >>>>> > >>>>> > Yesterday I moved our production system to that version. After a >>>>> few >>>>> > hours the problems started: stuck BLFs, hangups, LUA scripts >>>>> > sometimes do not run. I had to revert back to the old version. >>>>> > >>>>> > My question: Is there some simple test suite which I can test >>>>> > various scnarios on my development system? like, "call XX, while >>>>> > this call is active call YY which is monitored by XX and hangup", >>>>> > etc. It is hard to cover all cases manually, so I would like >>>>> > something automated. >>>>> > >>>>> > Thanks, __Yehavi: >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > -- >>>>> > Michael S Collins >>>>> > Twitter: @mercutioviz >>>>> > http://www.FreeSWITCH.org >>>>> > http://www.ClueCon.com >>>>> > http://www.OSTAG.org >>>>> > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Marcin Gozdalik >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/0dc81895/attachment-0001.html From gozdal at gmail.com Wed Jan 23 21:09:28 2013 From: gozdal at gmail.com (Marcin Gozdalik) Date: Wed, 23 Jan 2013 19:09:28 +0100 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: <50FE1857.7020804@communicatefreely.net> Message-ID: In general I think it's true, but maybe at lest some of the cases may be covered by some automated scripts? Like knowing that version X of firmware for Y brand phones requires certain header to be present or shorter than X bytes or whatever could provide automated test case for all future versions. Of course it won't cover new firmwares or new phones or some obscure cases but maybe at least most common cases of regressions in code could be caught using automated testing without phones. Personally I don't believe in "hook some Arduino to VoIP phone and check if some LED lights up" to be supported by open-source community. It might be possible inside commercial company that would keep the stack of VoIP phones and custom-made hardware tailored for those phones, but it won't leverage community folks. 2013/1/23 Michael Collins > After 4 years of working on the CudaTel team and hearing about SLA and BLF > failures I can tell you that you need to see what's actually displayed on > the phone itself, so checking the SIP traffic is not sufficient for a > thorough test. > > If anyone is willing to crack open a phone and do some wiring they might > be able to use a raspi or arduino and some components to monitor the phone. > It's not the easiest thing in the world to do. That's why having a phone > with an API would be so helpful. Autotesting would save enormous amounts of > time in the long run. > > -MC > > On Wed, Jan 23, 2013 at 4:18 AM, Regis M wrote: > >> Maybe It's could be done with some sip session record that we can >> playback to freeswitch ? >> >> In this mode, sofia will ignore or bypass time announce and sequence >> number to anwer the correct/same thing as the first "real" recording >> sequence. >> >> A sort of mod_sofia_test, one FS can read pcap or sip sequence and throw >> mod_sofia_test "simulate" the endpoint to another or itself. >> It could allow to record all phone session (register, makecall, holdcall, >> transfer,...) and check FS still answer same things or corrects thing. >> >> A little hard maybe ? :) >> I'm not a specialist of "sequence" or SIP change that occurs in a session >> dialog between 2 calls from exactly the same device. >> >> >> >> >> >> 2013/1/23 Marcin Gozdalik >> >>> I'd doubt there is any VoIP phone with such support. >>> >>> But I'd guess writing tests querying actual phones wouldn't be much more >>> work than: >>> a) testing what kind of messages for BLF/MWI actually work on given phone >>> b) testing (in ESL?) if FS sends good messages when certain actions are >>> done (leaving voicemail or making a call) >>> >>> >>> >>> >>> 2013/1/22 Michael Collins >>> >>>> That begs the question: does anyone know of any SIP phones that have an >>>> API where you can poll the status of the lamps as well as what's on the >>>> display, whether MWI is on, etc.? That would be epic. >>>> >>>> -MC >>>> >>>> On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre < >>>> fs-list at communicatefreely.net> wrote: >>>> >>>>> Hmm. Good idea. >>>>> >>>>> I know that you can do this with Aastra phones, and possibly others. >>>>> >>>>> With the Aastra's, there is an XML API that you can use to push >>>>> notifications, apps, etc. >>>>> You can push XML to the phone on an HTTP interface (there is an ACL >>>>> to tell it where to >>>>> allow it from), and you can initiate phone calls, as well as hang up, >>>>> transfer, etc. >>>>> There is also an action URI, so you can have the phone report what is >>>>> coming in on caller >>>>> ID, when it is ringing, hung up, etc. >>>>> >>>>> I don't know if you can get lamp status from this or not, but perhaps >>>>> you could have a >>>>> system where a pattern of calls is generated, and after many >>>>> repetitions, the system calls >>>>> or a human to check and see that the lights match an expected state. >>>>> >>>>> Just an idea anyway. >>>>> >>>>> -Tim >>>>> >>>>> Michael Collins wrote: >>>>> > This test suite does not presently exist. I would love to see some >>>>> work >>>>> > done in this area. >>>>> > >>>>> > One challenge is that there are numerous kinds of devices that can >>>>> > interop with FS so the test suite will need to have some way to >>>>> handle >>>>> > the various phone makes and models. Also, I don't know which hard >>>>> > phones, if any, support automated testing. I would *love* to see some >>>>> > sort of web/network API on various phones where you could poll the >>>>> phone >>>>> > to see the state of each SLA/BLF key as well as what's in the >>>>> display. >>>>> > That would make automated testing much cleaner. As it stands now, >>>>> > testing BLFs is a tedious process of watching phones and waiting to >>>>> see >>>>> > if the lights do what they should be doing. >>>>> > >>>>> > If anyone has practical suggestions (nothing pie-in-the-sky, if you >>>>> get >>>>> > my meaning) then by all means share with the group. >>>>> > >>>>> > -MC >>>>> > >>>>> > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine >>>>> > > >>>>> wrote: >>>>> > >>>>> > Hello, >>>>> > >>>>> > We are running version 1.0.6; we did not upgrade for a long >>>>> time >>>>> > due to some problems (mainly stuck BLFs) with the newer versions >>>>> > which have been corrected a few months ago. >>>>> > >>>>> > A few weeks ago I pulled the latest GIT and did functional >>>>> testings >>>>> > on our development system; all was OK. >>>>> > >>>>> > Yesterday I moved our production system to that version. After a >>>>> few >>>>> > hours the problems started: stuck BLFs, hangups, LUA scripts >>>>> > sometimes do not run. I had to revert back to the old version. >>>>> > >>>>> > My question: Is there some simple test suite which I can test >>>>> > various scnarios on my development system? like, "call XX, while >>>>> > this call is active call YY which is monitored by XX and hangup", >>>>> > etc. It is hard to cover all cases manually, so I would like >>>>> > something automated. >>>>> > >>>>> > Thanks, __Yehavi: >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > -- >>>>> > Michael S Collins >>>>> > Twitter: @mercutioviz >>>>> > http://www.FreeSWITCH.org >>>>> > http://www.ClueCon.com >>>>> > http://www.OSTAG.org >>>>> > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Marcin Gozdalik >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Marcin Gozdalik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/cd284c39/attachment-0001.html From jmesquita at freeswitch.org Wed Jan 23 21:32:37 2013 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 23 Jan 2013 15:32:37 -0300 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: <50FE1857.7020804@communicatefreely.net> Message-ID: <061B51D7-26EC-42EC-9E66-B74A78E92F42@freeswitch.org> Mc, agree to disagree. That is what is so wonderful in this brave new world of Ngn telephony of ours. If you limit the number of devices you declare compatible with your platform and have the money to test each and every phone of that list on every and each firmware it has, your thorough test would be feasible and ONLY If the list of devices is small. But if you have a long list of devices you're screwed and instead of developing the product itself you'll spend your time testing phones or even worse, making patches for each model that works differently. I guess that all there is to do is stick to standard compliance and report to manufacturers when thy are not so they can fix their crap. Even that is not enough, I know, because there is always a customer saying: " but it works with Asterisk...." Anyhow, a sip compliance test would be awesome but really hard to accomplish correctly because it implies almost implementing another stack (kinda). Sent from my iPhone On Jan 23, 2013, at 2:25 PM, Michael Collins wrote: > After 4 years of working on the CudaTel team and hearing about SLA and BLF failures I can tell you that you need to see what's actually displayed on the phone itself, so checking the SIP traffic is not sufficient for a thorough test. > > If anyone is willing to crack open a phone and do some wiring they might be able to use a raspi or arduino and some components to monitor the phone. It's not the easiest thing in the world to do. That's why having a phone with an API would be so helpful. Autotesting would save enormous amounts of time in the long run. > > -MC > > On Wed, Jan 23, 2013 at 4:18 AM, Regis M wrote: >> Maybe It's could be done with some sip session record that we can playback to freeswitch ? >> >> In this mode, sofia will ignore or bypass time announce and sequence number to anwer the correct/same thing as the first "real" recording sequence. >> >> A sort of mod_sofia_test, one FS can read pcap or sip sequence and throw mod_sofia_test "simulate" the endpoint to another or itself. >> It could allow to record all phone session (register, makecall, holdcall, transfer,...) and check FS still answer same things or corrects thing. >> >> A little hard maybe ? :) >> I'm not a specialist of "sequence" or SIP change that occurs in a session dialog between 2 calls from exactly the same device. >> >> >> >> >> >> 2013/1/23 Marcin Gozdalik >>> I'd doubt there is any VoIP phone with such support. >>> >>> But I'd guess writing tests querying actual phones wouldn't be much more work than: >>> a) testing what kind of messages for BLF/MWI actually work on given phone >>> b) testing (in ESL?) if FS sends good messages when certain actions are done (leaving voicemail or making a call) >>> >>> >>> >>> >>> 2013/1/22 Michael Collins >>>> That begs the question: does anyone know of any SIP phones that have an API where you can poll the status of the lamps as well as what's on the display, whether MWI is on, etc.? That would be epic. >>>> >>>> -MC >>>> >>>> On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre wrote: >>>>> Hmm. Good idea. >>>>> >>>>> I know that you can do this with Aastra phones, and possibly others. >>>>> >>>>> With the Aastra's, there is an XML API that you can use to push notifications, apps, etc. >>>>> You can push XML to the phone on an HTTP interface (there is an ACL to tell it where to >>>>> allow it from), and you can initiate phone calls, as well as hang up, transfer, etc. >>>>> There is also an action URI, so you can have the phone report what is coming in on caller >>>>> ID, when it is ringing, hung up, etc. >>>>> >>>>> I don't know if you can get lamp status from this or not, but perhaps you could have a >>>>> system where a pattern of calls is generated, and after many repetitions, the system calls >>>>> or a human to check and see that the lights match an expected state. >>>>> >>>>> Just an idea anyway. >>>>> >>>>> -Tim >>>>> >>>>> Michael Collins wrote: >>>>> > This test suite does not presently exist. I would love to see some work >>>>> > done in this area. >>>>> > >>>>> > One challenge is that there are numerous kinds of devices that can >>>>> > interop with FS so the test suite will need to have some way to handle >>>>> > the various phone makes and models. Also, I don't know which hard >>>>> > phones, if any, support automated testing. I would *love* to see some >>>>> > sort of web/network API on various phones where you could poll the phone >>>>> > to see the state of each SLA/BLF key as well as what's in the display. >>>>> > That would make automated testing much cleaner. As it stands now, >>>>> > testing BLFs is a tedious process of watching phones and waiting to see >>>>> > if the lights do what they should be doing. >>>>> > >>>>> > If anyone has practical suggestions (nothing pie-in-the-sky, if you get >>>>> > my meaning) then by all means share with the group. >>>>> > >>>>> > -MC >>>>> > >>>>> > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine >>>>> > > wrote: >>>>> > >>>>> > Hello, >>>>> > >>>>> > We are running version 1.0.6; we did not upgrade for a long time >>>>> > due to some problems (mainly stuck BLFs) with the newer versions >>>>> > which have been corrected a few months ago. >>>>> > >>>>> > A few weeks ago I pulled the latest GIT and did functional testings >>>>> > on our development system; all was OK. >>>>> > >>>>> > Yesterday I moved our production system to that version. After a few >>>>> > hours the problems started: stuck BLFs, hangups, LUA scripts >>>>> > sometimes do not run. I had to revert back to the old version. >>>>> > >>>>> > My question: Is there some simple test suite which I can test >>>>> > various scnarios on my development system? like, "call XX, while >>>>> > this call is active call YY which is monitored by XX and hangup", >>>>> > etc. It is hard to cover all cases manually, so I would like >>>>> > something automated. >>>>> > >>>>> > Thanks, __Yehavi: >>>>> > >>>>> > _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > -- >>>>> > Michael S Collins >>>>> > Twitter: @mercutioviz >>>>> > http://www.FreeSWITCH.org >>>>> > http://www.ClueCon.com >>>>> > http://www.OSTAG.org >>>>> > >>>>> > >>>>> > ------------------------------------------------------------------------ >>>>> > >>>>> > _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> -- >>> Marcin Gozdalik >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/b6f96796/attachment-0001.html From cingusoft at gmail.com Wed Jan 23 21:16:34 2013 From: cingusoft at gmail.com (Andrea Mucci) Date: Wed, 23 Jan 2013 19:16:34 +0100 Subject: [Freeswitch-users] FreeSwitch Multitenant registrar and generic SIP PBX Message-ID: Hi All I have a project for one university, this is a free project and i want to help him to solve some thinks. the idea is create a sytem multitenant or multicompany for this university. the IP PBX work with sip and don't have any multitenant service. My idea is use the Multitenat service of Fresswitch and all other services like RTP media etc with the IP PBX. Is this possible? The schema will be., UA send a register "(1000 at test.com) to Freeswitch Freeswitch register with the SIP PBX with the extension 2000. calls and services are mediated by the SIP PBX. UA send a register "1001 at test.co to freeswith Freeswitch send a register request to the SIP PBX to the 2001 extension calls and services are mediated by the SIP pBX sorry if you tink this is a supid question but i'm newbie and i want to help this university with this project. have a nice day -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/0f128f46/attachment.html From julian.pawlowski at gmail.com Wed Jan 23 21:19:54 2013 From: julian.pawlowski at gmail.com (Julian Pawlowski) Date: Wed, 23 Jan 2013 19:19:54 +0100 Subject: [Freeswitch-users] Debian unixODBC issues Message-ID: Hi all, as mentioned in http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#PostgreSQL_8.4_on_Debian_Squeezethere seem to be some stability issues with the unixODBC package coming with Debian. I heard this is still valid for Wheezy too so I was just contacting the package maintainer who was quite surprised to hear about this issue. Obviously nobody told him before which is why Wheezy still uses unixODBC version 2.2. Now that we have a freeze for Wheezy to prepare for the official release unixODBC 2.3 can't be included anymore. However version 2.2 could get a patch if we are able to provide enough information about the root cause. So this is a chance for FreeSwitch's own packages to work really smooth with Debian without any manual correction needed ;-) Would anybody be able to provide more advanced information, for instance a backtrace and instructions on how to reproduce the issue? Cheers, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/52df99c5/attachment.html From avi at avimarcus.net Wed Jan 23 21:57:54 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 23 Jan 2013 20:57:54 +0200 Subject: [Freeswitch-users] Debian unixODBC issues In-Reply-To: References: Message-ID: The link has Postgres in the hashtag. Does that mean you're specifically interested in postgres? If so, that's now accessible directly via freeswitch without odbc middleware. -Avi On Wed, Jan 23, 2013 at 8:19 PM, Julian Pawlowski < julian.pawlowski at gmail.com> wrote: > Hi all, > > as mentioned in > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#PostgreSQL_8.4_on_Debian_Squeezethere seem to be some stability issues with the unixODBC package coming > with Debian. > > I heard this is still valid for Wheezy too so I was just contacting the > package maintainer who was quite surprised to hear about this issue. > Obviously nobody told him before which is why Wheezy still uses unixODBC > version 2.2. > > Now that we have a freeze for Wheezy to prepare for the official release > unixODBC 2.3 can't be included anymore. However version 2.2 could get a > patch if we are able to provide enough information about the root cause. So > this is a chance for FreeSwitch's own packages to work really smooth with > Debian without any manual correction needed ;-) > > Would anybody be able to provide more advanced information, for instance a > backtrace and instructions on how to reproduce the issue? > > > Cheers, > Julian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/a4e0e0c7/attachment.html From msc at freeswitch.org Wed Jan 23 21:59:51 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Jan 2013 10:59:51 -0800 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: <061B51D7-26EC-42EC-9E66-B74A78E92F42@freeswitch.org> References: <50FE1857.7020804@communicatefreely.net> <061B51D7-26EC-42EC-9E66-B74A78E92F42@freeswitch.org> Message-ID: Well, there's a difference between testing to certify as compliant and testing to see if the latest version of FS breaks something. In any event, I think Avi's suggestion about a script (or whatever) that runs while a human sits and watches a phone. It could even be interactive: What extension number is the test phone? >2000 Sending call to 2000. Is the phone ringing? (Y/N) >Y Is the line key blinking? (Y/N) >Y Interactivity is not a requirement, just an option. It could even just spit out the name of the test that it's running and offer some information about what it thinks should be happening on the phone at the moment. -MC On Wed, Jan 23, 2013 at 10:32 AM, Jo?o Mesquita wrote: > Mc, agree to disagree. That is what is so wonderful in this brave new > world of Ngn telephony of ours. If you limit the number of devices you > declare compatible with your platform and have the money to test each and > every phone of that list on every and each firmware it has, your thorough > test would be feasible and ONLY If the list of devices is small. > > But if you have a long list of devices you're screwed and instead of > developing the product itself you'll spend your time testing phones or even > worse, making patches for each model that works differently. > > I guess that all there is to do is stick to standard compliance and report > to manufacturers when thy are not so they can fix their crap. Even that is > not enough, I know, because there is always a customer saying: " but it > works with Asterisk...." > > Anyhow, a sip compliance test would be awesome but really hard to > accomplish correctly because it implies almost implementing another stack > (kinda). > > Sent from my iPhone > > On Jan 23, 2013, at 2:25 PM, Michael Collins wrote: > > After 4 years of working on the CudaTel team and hearing about SLA and BLF > failures I can tell you that you need to see what's actually displayed on > the phone itself, so checking the SIP traffic is not sufficient for a > thorough test. > > If anyone is willing to crack open a phone and do some wiring they might > be able to use a raspi or arduino and some components to monitor the phone. > It's not the easiest thing in the world to do. That's why having a phone > with an API would be so helpful. Autotesting would save enormous amounts of > time in the long run. > > -MC > > On Wed, Jan 23, 2013 at 4:18 AM, Regis M wrote: > >> Maybe It's could be done with some sip session record that we can >> playback to freeswitch ? >> >> In this mode, sofia will ignore or bypass time announce and sequence >> number to anwer the correct/same thing as the first "real" recording >> sequence. >> >> A sort of mod_sofia_test, one FS can read pcap or sip sequence and throw >> mod_sofia_test "simulate" the endpoint to another or itself. >> It could allow to record all phone session (register, makecall, holdcall, >> transfer,...) and check FS still answer same things or corrects thing. >> >> A little hard maybe ? :) >> I'm not a specialist of "sequence" or SIP change that occurs in a session >> dialog between 2 calls from exactly the same device. >> >> >> >> >> >> 2013/1/23 Marcin Gozdalik >> >>> I'd doubt there is any VoIP phone with such support. >>> >>> But I'd guess writing tests querying actual phones wouldn't be much more >>> work than: >>> a) testing what kind of messages for BLF/MWI actually work on given phone >>> b) testing (in ESL?) if FS sends good messages when certain actions are >>> done (leaving voicemail or making a call) >>> >>> >>> >>> >>> 2013/1/22 Michael Collins >>> >>>> That begs the question: does anyone know of any SIP phones that have an >>>> API where you can poll the status of the lamps as well as what's on the >>>> display, whether MWI is on, etc.? That would be epic. >>>> >>>> -MC >>>> >>>> On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre < >>>> fs-list at communicatefreely.net> wrote: >>>> >>>>> Hmm. Good idea. >>>>> >>>>> I know that you can do this with Aastra phones, and possibly others. >>>>> >>>>> With the Aastra's, there is an XML API that you can use to push >>>>> notifications, apps, etc. >>>>> You can push XML to the phone on an HTTP interface (there is an ACL >>>>> to tell it where to >>>>> allow it from), and you can initiate phone calls, as well as hang up, >>>>> transfer, etc. >>>>> There is also an action URI, so you can have the phone report what is >>>>> coming in on caller >>>>> ID, when it is ringing, hung up, etc. >>>>> >>>>> I don't know if you can get lamp status from this or not, but perhaps >>>>> you could have a >>>>> system where a pattern of calls is generated, and after many >>>>> repetitions, the system calls >>>>> or a human to check and see that the lights match an expected state. >>>>> >>>>> Just an idea anyway. >>>>> >>>>> -Tim >>>>> >>>>> Michael Collins wrote: >>>>> > This test suite does not presently exist. I would love to see some >>>>> work >>>>> > done in this area. >>>>> > >>>>> > One challenge is that there are numerous kinds of devices that can >>>>> > interop with FS so the test suite will need to have some way to >>>>> handle >>>>> > the various phone makes and models. Also, I don't know which hard >>>>> > phones, if any, support automated testing. I would *love* to see some >>>>> > sort of web/network API on various phones where you could poll the >>>>> phone >>>>> > to see the state of each SLA/BLF key as well as what's in the >>>>> display. >>>>> > That would make automated testing much cleaner. As it stands now, >>>>> > testing BLFs is a tedious process of watching phones and waiting to >>>>> see >>>>> > if the lights do what they should be doing. >>>>> > >>>>> > If anyone has practical suggestions (nothing pie-in-the-sky, if you >>>>> get >>>>> > my meaning) then by all means share with the group. >>>>> > >>>>> > -MC >>>>> > >>>>> > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine >>>>> > > >>>>> wrote: >>>>> > >>>>> > Hello, >>>>> > >>>>> > We are running version 1.0.6; we did not upgrade for a long >>>>> time >>>>> > due to some problems (mainly stuck BLFs) with the newer versions >>>>> > which have been corrected a few months ago. >>>>> > >>>>> > A few weeks ago I pulled the latest GIT and did functional >>>>> testings >>>>> > on our development system; all was OK. >>>>> > >>>>> > Yesterday I moved our production system to that version. After a >>>>> few >>>>> > hours the problems started: stuck BLFs, hangups, LUA scripts >>>>> > sometimes do not run. I had to revert back to the old version. >>>>> > >>>>> > My question: Is there some simple test suite which I can test >>>>> > various scnarios on my development system? like, "call XX, while >>>>> > this call is active call YY which is monitored by XX and hangup", >>>>> > etc. It is hard to cover all cases manually, so I would like >>>>> > something automated. >>>>> > >>>>> > Thanks, __Yehavi: >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > -- >>>>> > Michael S Collins >>>>> > Twitter: @mercutioviz >>>>> > http://www.FreeSWITCH.org >>>>> > http://www.ClueCon.com >>>>> > http://www.OSTAG.org >>>>> > >>>>> > >>>>> > >>>>> ------------------------------------------------------------------------ >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Marcin Gozdalik >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/0732331c/attachment-0001.html From moises.silva at gmail.com Wed Jan 23 22:00:10 2013 From: moises.silva at gmail.com (Moises Silva) Date: Wed, 23 Jan 2013 14:00:10 -0500 Subject: [Freeswitch-users] bluetooth endpoint In-Reply-To: References: <20130120235956.69310@gmx.com> Message-ID: On Mon, Jan 21, 2013 at 12:01 AM, Dmitry Lysenko wrote: > Only gsmopen exist in freeswitch at this moment for this purposes, but it > does not support bluetooth link with cell phones. > > Well, that's not quite true. I did a mod_handsfree (which is Bluetooth) module for FreeSWITCH a year or two ago. It's a hack, pretty rough, but I was able to receive and place calls from my iPhone with it. https://github.com/moises-silva/mod_handsfree I never submitted this to the standard FreeSWITCH tree because it's kind of hackish, it uses D-BUS, which is a pain to write in C, so it uses some helper python scripts that are called by FreeSWITCH for call control. Also requires a recent version of D-BUS which CentOS does not have, I believe I only found it available under Arch Linux. *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/da1c6986/attachment.html From anton.jugatsu at gmail.com Wed Jan 23 22:10:21 2013 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Wed, 23 Jan 2013 23:10:21 +0400 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today In-Reply-To: References: Message-ID: Michael, correct me if I'm wrong but there ain't any slides at cluecon.com. 2013/1/23 Michael Collins > Hello folks, > > On today's call we have a few news items to discuss, after which we will > have an open forum for Q&A on FreeSWITCH topics. The agenda is here: > > http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_23 > > Talk to you soon! > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/12d343ec/attachment.html From avi at avimarcus.net Wed Jan 23 22:14:35 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 23 Jan 2013 21:14:35 +0200 Subject: [Freeswitch-users] How to test Freeswitch before moving to production? In-Reply-To: References: <50FE1857.7020804@communicatefreely.net> <061B51D7-26EC-42EC-9E66-B74A78E92F42@freeswitch.org> Message-ID: Taking a step back -- much of this can be monitored simply by watching FS events, especially if we have dummy/cooperative endpoints for testing. E.g. when you run "originate $long_bridge_with_vars" does a new channel get started? etc. Individual phone compliance is a different issue. -Avi On Wed, Jan 23, 2013 at 8:59 PM, Michael Collins wrote: > Well, there's a difference between testing to certify as compliant and > testing to see if the latest version of FS breaks something. In any event, > I think Avi's suggestion about a script (or whatever) that runs while a > human sits and watches a phone. It could even be interactive: > > What extension number is the test phone? > >2000 > > Sending call to 2000. Is the phone ringing? (Y/N) > >Y > > Is the line key blinking? (Y/N) > >Y > > Interactivity is not a requirement, just an option. It could even just > spit out the name of the test that it's running and offer some information > about what it thinks should be happening on the phone at the moment. > > -MC > > > On Wed, Jan 23, 2013 at 10:32 AM, Jo?o Mesquita wrote: > >> Mc, agree to disagree. That is what is so wonderful in this brave new >> world of Ngn telephony of ours. If you limit the number of devices you >> declare compatible with your platform and have the money to test each and >> every phone of that list on every and each firmware it has, your thorough >> test would be feasible and ONLY If the list of devices is small. >> >> But if you have a long list of devices you're screwed and instead of >> developing the product itself you'll spend your time testing phones or even >> worse, making patches for each model that works differently. >> >> I guess that all there is to do is stick to standard compliance and >> report to manufacturers when thy are not so they can fix their crap. Even >> that is not enough, I know, because there is always a customer saying: " >> but it works with Asterisk...." >> >> Anyhow, a sip compliance test would be awesome but really hard to >> accomplish correctly because it implies almost implementing another stack >> (kinda). >> >> Sent from my iPhone >> >> On Jan 23, 2013, at 2:25 PM, Michael Collins wrote: >> >> After 4 years of working on the CudaTel team and hearing about SLA and >> BLF failures I can tell you that you need to see what's actually displayed >> on the phone itself, so checking the SIP traffic is not sufficient for a >> thorough test. >> >> If anyone is willing to crack open a phone and do some wiring they might >> be able to use a raspi or arduino and some components to monitor the phone. >> It's not the easiest thing in the world to do. That's why having a phone >> with an API would be so helpful. Autotesting would save enormous amounts of >> time in the long run. >> >> -MC >> >> On Wed, Jan 23, 2013 at 4:18 AM, Regis M > > wrote: >> >>> Maybe It's could be done with some sip session record that we can >>> playback to freeswitch ? >>> >>> In this mode, sofia will ignore or bypass time announce and sequence >>> number to anwer the correct/same thing as the first "real" recording >>> sequence. >>> >>> A sort of mod_sofia_test, one FS can read pcap or sip sequence and throw >>> mod_sofia_test "simulate" the endpoint to another or itself. >>> It could allow to record all phone session (register, makecall, >>> holdcall, transfer,...) and check FS still answer same things or corrects >>> thing. >>> >>> A little hard maybe ? :) >>> I'm not a specialist of "sequence" or SIP change that occurs in a >>> session dialog between 2 calls from exactly the same device. >>> >>> >>> >>> >>> >>> 2013/1/23 Marcin Gozdalik >>> >>>> I'd doubt there is any VoIP phone with such support. >>>> >>>> But I'd guess writing tests querying actual phones wouldn't be much >>>> more work than: >>>> a) testing what kind of messages for BLF/MWI actually work on given >>>> phone >>>> b) testing (in ESL?) if FS sends good messages when certain actions are >>>> done (leaving voicemail or making a call) >>>> >>>> >>>> >>>> >>>> 2013/1/22 Michael Collins >>>> >>>>> That begs the question: does anyone know of any SIP phones that have >>>>> an API where you can poll the status of the lamps as well as what's on the >>>>> display, whether MWI is on, etc.? That would be epic. >>>>> >>>>> -MC >>>>> >>>>> On Mon, Jan 21, 2013 at 8:40 PM, Tim St. Pierre < >>>>> fs-list at communicatefreely.net> wrote: >>>>> >>>>>> Hmm. Good idea. >>>>>> >>>>>> I know that you can do this with Aastra phones, and possibly others. >>>>>> >>>>>> With the Aastra's, there is an XML API that you can use to push >>>>>> notifications, apps, etc. >>>>>> You can push XML to the phone on an HTTP interface (there is an ACL >>>>>> to tell it where to >>>>>> allow it from), and you can initiate phone calls, as well as hang up, >>>>>> transfer, etc. >>>>>> There is also an action URI, so you can have the phone report what is >>>>>> coming in on caller >>>>>> ID, when it is ringing, hung up, etc. >>>>>> >>>>>> I don't know if you can get lamp status from this or not, but perhaps >>>>>> you could have a >>>>>> system where a pattern of calls is generated, and after many >>>>>> repetitions, the system calls >>>>>> or a human to check and see that the lights match an expected state. >>>>>> >>>>>> Just an idea anyway. >>>>>> >>>>>> -Tim >>>>>> >>>>>> Michael Collins wrote: >>>>>> > This test suite does not presently exist. I would love to see some >>>>>> work >>>>>> > done in this area. >>>>>> > >>>>>> > One challenge is that there are numerous kinds of devices that can >>>>>> > interop with FS so the test suite will need to have some way to >>>>>> handle >>>>>> > the various phone makes and models. Also, I don't know which hard >>>>>> > phones, if any, support automated testing. I would *love* to see >>>>>> some >>>>>> > sort of web/network API on various phones where you could poll the >>>>>> phone >>>>>> > to see the state of each SLA/BLF key as well as what's in the >>>>>> display. >>>>>> > That would make automated testing much cleaner. As it stands now, >>>>>> > testing BLFs is a tedious process of watching phones and waiting to >>>>>> see >>>>>> > if the lights do what they should be doing. >>>>>> > >>>>>> > If anyone has practical suggestions (nothing pie-in-the-sky, if you >>>>>> get >>>>>> > my meaning) then by all means share with the group. >>>>>> > >>>>>> > -MC >>>>>> > >>>>>> > On Mon, Jan 21, 2013 at 10:03 AM, Yehavi Bourvine >>>>>> > > >>>>>> wrote: >>>>>> > >>>>>> > Hello, >>>>>> > >>>>>> > We are running version 1.0.6; we did not upgrade for a long >>>>>> time >>>>>> > due to some problems (mainly stuck BLFs) with the newer versions >>>>>> > which have been corrected a few months ago. >>>>>> > >>>>>> > A few weeks ago I pulled the latest GIT and did functional >>>>>> testings >>>>>> > on our development system; all was OK. >>>>>> > >>>>>> > Yesterday I moved our production system to that version. After >>>>>> a few >>>>>> > hours the problems started: stuck BLFs, hangups, LUA scripts >>>>>> > sometimes do not run. I had to revert back to the old version. >>>>>> > >>>>>> > My question: Is there some simple test suite which I can test >>>>>> > various scnarios on my development system? like, "call XX, while >>>>>> > this call is active call YY which is monitored by XX and >>>>>> hangup", >>>>>> > etc. It is hard to cover all cases manually, so I would like >>>>>> > something automated. >>>>>> > >>>>>> > Thanks, __Yehavi: >>>>>> > >>>>>> > >>>>>> _________________________________________________________________________ >>>>>> > Professional FreeSWITCH Consulting Services: >>>>>> > consulting at freeswitch.org >>>>>> > http://www.freeswitchsolutions.com >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > Official FreeSWITCH Sites >>>>>> > http://www.freeswitch.org >>>>>> > http://wiki.freeswitch.org >>>>>> > http://www.cluecon.com >>>>>> > >>>>>> > FreeSWITCH-users mailing list >>>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>>> > >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > -- >>>>>> > Michael S Collins >>>>>> > Twitter: @mercutioviz >>>>>> > http://www.FreeSWITCH.org >>>>>> > http://www.ClueCon.com >>>>>> > http://www.OSTAG.org >>>>>> > >>>>>> > >>>>>> > >>>>>> ------------------------------------------------------------------------ >>>>>> > >>>>>> > >>>>>> _________________________________________________________________________ >>>>>> > Professional FreeSWITCH Consulting Services: >>>>>> > consulting at freeswitch.org >>>>>> > http://www.freeswitchsolutions.com >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > Official FreeSWITCH Sites >>>>>> > http://www.freeswitch.org >>>>>> > http://wiki.freeswitch.org >>>>>> > http://www.cluecon.com >>>>>> > >>>>>> > FreeSWITCH-users mailing list >>>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Michael S Collins >>>>> Twitter: @mercutioviz >>>>> http://www.FreeSWITCH.org >>>>> http://www.ClueCon.com >>>>> http://www.OSTAG.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Marcin Gozdalik >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/9b0ba995/attachment-0001.html From msc at freeswitch.org Wed Jan 23 22:30:10 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Jan 2013 11:30:10 -0800 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today In-Reply-To: References: Message-ID: Click on the "2013" and you'll see a drop-down. Select "2012" and then click "Schedule". From there click on the name of the talk you want to see. -MC On Wed, Jan 23, 2013 at 11:10 AM, Anton Kvashenkin wrote: > Michael, correct me if I'm wrong but there ain't any slides at cluecon.com > . > > > 2013/1/23 Michael Collins > >> Hello folks, >> >> On today's call we have a few news items to discuss, after which we will >> have an open forum for Q&A on FreeSWITCH topics. The agenda is here: >> >> http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_23 >> >> Talk to you soon! >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/a3d2f807/attachment.html From anton.jugatsu at gmail.com Wed Jan 23 22:47:07 2013 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Wed, 23 Jan 2013 23:47:07 +0400 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today In-Reply-To: References: Message-ID: Thanks. 2013/1/23 Michael Collins > Click on the "2013" and you'll see a drop-down. Select "2012" and then > click "Schedule". From there click on the name of the talk you want to see. > > -MC > > > On Wed, Jan 23, 2013 at 11:10 AM, Anton Kvashenkin < > anton.jugatsu at gmail.com> wrote: > >> Michael, correct me if I'm wrong but there ain't any slides at >> cluecon.com. >> >> >> 2013/1/23 Michael Collins >> >>> Hello folks, >>> >>> On today's call we have a few news items to discuss, after which we will >>> have an open forum for Q&A on FreeSWITCH topics. The agenda is here: >>> >>> http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_23 >>> >>> Talk to you soon! >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/6161a072/attachment.html From mail at loredo.me Wed Jan 23 23:08:15 2013 From: mail at loredo.me (Julian Pawlowski) Date: Wed, 23 Jan 2013 21:08:15 +0100 Subject: [Freeswitch-users] Debian unixODBC issues In-Reply-To: References: Message-ID: Hi Avi, On Wed, Jan 23, 2013 at 7:57 PM, Avi Marcus wrote: > The link has Postgres in the hashtag. Does that mean you're specifically > interested in postgres? > No. Postgres is only an example (in fact we are using MySQL as of now). > If so, that's now accessible directly via freeswitch without odbc > middleware. > I know. My email is not regarding to to run FreeSwitch with a database directly or via ODBC middleware, it's about the bug that was noted there: "*We've experienced some major stability issues with the unixODBC-debian-package 2.2.14p2-1 from the repository. FreeSWITCH would die anywhere between a few days to weeks. Upgrading to a manually compiled unixODBC 2.3.0-pre fixed this problem and has been tested stable for the last 2 months. --Peletiah 13:04, 4 February 2012 (UTC)"* Now re-read my message with new glases :-) Cheers, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/ffd05e1b/attachment.html From cmason at frontiernetworks.ca Wed Jan 23 23:14:46 2013 From: cmason at frontiernetworks.ca (Colin Mason) Date: Wed, 23 Jan 2013 15:14:46 -0500 Subject: [Freeswitch-users] Debian unixODBC issues In-Reply-To: References: Message-ID: <0D1C698866F66045A6201FD0F59CAC90014716C4E8@EX.frontier.local> I had stability issues with unixODBC-2.2.14-12 on CentOS 6. Adding Threading = 0 to my /etc/odbcinst.ini fixed the issues I was having. Colin From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Julian Pawlowski Sent: Wednesday, January 23, 2013 3:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Debian unixODBC issues Hi Avi, On Wed, Jan 23, 2013 at 7:57 PM, Avi Marcus > wrote: The link has Postgres in the hashtag. Does that mean you're specifically interested in postgres? No. Postgres is only an example (in fact we are using MySQL as of now). If so, that's now accessible directly via freeswitch without odbc middleware. I know. My email is not regarding to to run FreeSwitch with a database directly or via ODBC middleware, it's about the bug that was noted there: "We've experienced some major stability issues with the unixODBC-debian-package 2.2.14p2-1 from the repository. FreeSWITCH would die anywhere between a few days to weeks. Upgrading to a manually compiled unixODBC 2.3.0-pre fixed this problem and has been tested stable for the last 2 months. --Peletiah 13:04, 4 February 2012 (UTC)" Now re-read my message with new glases :-) Cheers, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/11b39f9b/attachment-0001.html From steveayre at gmail.com Wed Jan 23 23:20:23 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 23 Jan 2013 20:20:23 +0000 Subject: [Freeswitch-users] Debian unixODBC issues In-Reply-To: References: Message-ID: Personally I'm using FS + unixODBC + myODBC for MySQL with no stability issues. It should be fine. If you do get issues report them on the Debian BTS - I'm sure they'd get it fixed, either by an updated package or backporting a fix to the current version. Bugs only get fixed if they're reported to the correct place people! The only time I have had problems was last week - it turned out I had a Percona APT repository installed and accidentally had installed a different libmysqlclient18 from the one that myodbc was compiled against (that one connected fine but did a segfault on any query). Perhaps Peletiah had some such problem, or was using a driver other than myODBC. In any case I don't think you'll find problems with myODBC, but obviously test before rolling into production. Wheezy is still 'testing' and shouldn't be on a production machine yet anyway, if you need newer packages check squeeze-backports ( http://backports.debian.org/) first. -Steve On 23 January 2013 20:08, Julian Pawlowski wrote: > Hi Avi, > > On Wed, Jan 23, 2013 at 7:57 PM, Avi Marcus wrote: > >> The link has Postgres in the hashtag. Does that mean you're specifically >> interested in postgres? >> > > No. Postgres is only an example (in fact we are using MySQL as of now). > > >> If so, that's now accessible directly via freeswitch without odbc >> middleware. >> > > I know. My email is not regarding to to run FreeSwitch with a database > directly or via ODBC middleware, it's about the bug that was noted there: > "*We've experienced some major stability issues with the > unixODBC-debian-package 2.2.14p2-1 from the repository. FreeSWITCH would > die anywhere between a few days to weeks. Upgrading to a manually compiled > unixODBC 2.3.0-pre fixed this problem and has been tested stable for the > last 2 months. --Peletiah 13:04, > 4 February 2012 (UTC)"* > > Now re-read my message with new glases :-) > > > Cheers, > Julian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/a10474bf/attachment.html From steveayre at gmail.com Wed Jan 23 23:22:43 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 23 Jan 2013 20:22:43 +0000 Subject: [Freeswitch-users] mod_stress In-Reply-To: <304825.1523.1358963241421.JavaMail.javamailuser@localhost> References: <18262686.1519.1358963234873.JavaMail.javamailuser@localhost> <304825.1523.1358963241421.JavaMail.javamailuser@localhost> Message-ID: It requires transcoding so it will increase CPU usage. That'll only increase CPU usage if you're not already transcoding. It also calculates the FFT, which from memory is a relatively expensive operation... there will be a noticable impact of using it. As for G729 licenses, it'll use an extra license for a channel that's not being transcoded/recorded/eavesdropped etc - but won't if you're already doing one of those on it. -Steve On 23 January 2013 17:47, Raimund Sacherer wrote: > Hi Peter, > > can you also make some statistics available on how much this module, if > activated, adds to the ressource usage? If you have a busy callcenter, > would this make a noticable impact on CPU usage? > > Also, would this use an additional G729 codec for calls using this codec? > > thanks, > best regards > Raimund > > ------------------------------ > *From: *"Peter Steinbach" > *To: *"FreeSWITCH Users Help" > *Sent: *Mi?rcoles, 23 de Enero 2013 0:34:41 > *Subject: *Re: [Freeswitch-users] mod_stress > > > Thanks Steven, > > I got it working and reported my howto and findings in the wiki: > http://wiki.freeswitch.org/wiki/Mod_stress > > I found it to be a nice application, but as you explained, there are some > false positives. > > Best regards > Peter > > On 01/22/13 12:39, Steven Ayre wrote: > > Media bugs are used to hook into the call and provide decoded audio to the > module. That requires transcoding, naturally. See stress_callback which is > called for each audio frame plus some init/cleanup hooks, and read up on > the API at http://docs.freeswitch.org/group__mb1.html. > > I suspect read vs write would correspond to the speaking/listening > directions of the channel. Read from caller, write to caller? The names > come from the media bug API. The read/write options will add the > corresponding media bug, stop will remove them. > > SWITCH_ADD_APP tells you it's a dialplan application - it's started from > dialplan, although you can also trigger dialplan apps on a channel via ESL > using uuid_broadcast. > > The FFT algorithm is used to extract the frequencies present in the > sample (the power spectrum). The algorithm then tries to detect stressed > based on that. Looks like the algorithm at least in part attempts to look > for tremor in the voice. More tremor indicates more stress. Over a > threshold it raises an event. > > Bear in mind that as it's pattern recognition (and little tested at > that) you'll get a lot of false positives and false negatives. It's also > looking for something in the voice that may or may not be there - some > people might be able to speak with an unstressed voice when they're > actually quite stressed. > > -Steve > > > > > On 22 January 2013 11:19, Peter Steinbach wrote: > >> Hello Anthony, >> >> although it may be incomplete, it may be useful for us. Can you give us >> some basic explanation, how this works? >> I can see in the code that there is a stress command with read|write|stop >> parameter. >> Is this to be issued via event socket or is it a dialplan application? I >> suspect that "read" somehow starts the detection and "stop" somehow stops >> the detection. For "write" I currently have no idea. >> >> I would be happy to test this and report my configuration, explanations >> and examples in the wiki. >> >> Best regards >> Peter >> >> >> >> On 01/22/13 00:30, Anthony Minessale wrote: >> >> That module is incomplete we made it several years ago to test some code >> that was supposed to pick up on stress in your voice in a media bug so we >> could fire events and whisper "lie" to the caller as he talked to the guy. >> >> >> >> On Mon, Jan 21, 2013 at 4:56 PM, Peter Steinbach wrote: >> >>> I am also wondering, what this module does in detail. >>> >>> Regarding the source code, I can see that FFT's are made in order to to >>> get some sort of frequency profile. >>> But the code and remarks do not give me a clue how this module actually >>> works. >>> >>> Best regards Peter >>> >>> >>> >>> On 05/26/12 15:05, BF wrote: >>> > Hello, >>> > >>> > I'd like to know what the "stress write" from mod_stress module does. >>> I have hard time finding documentation or an example about mod_stress >>> module. >>> > >>> > Anyone? >>> > >>> > Thank you. >>> > >>> > Eric >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > Join Us At ClueCon - Aug 7-9, 2012 >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> -- >>> With kind regards >>> Peter Steinbach >>> >>> Telefaks Services GmbH >>> mailto:lists (att) telefaks.de >>> Internet: www.telefaks.de >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> -- >> With kind regards >> Peter Steinbach >> >> Telefaks Services GmbHmailto:lists (att) telefaks.de >> Internet: www.telefaks.de >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbHmailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/87bcebb1/attachment-0001.html From mail at loredo.me Wed Jan 23 23:33:58 2013 From: mail at loredo.me (Julian Pawlowski) Date: Wed, 23 Jan 2013 21:33:58 +0100 Subject: [Freeswitch-users] Debian unixODBC issues In-Reply-To: References: Message-ID: Hi Steve, On Wed, Jan 23, 2013 at 9:20 PM, Steven Ayre wrote: > Personally I'm using FS + unixODBC + myODBC for MySQL with no stability > issues. It should be fine. The issue occurs under higher loads or peaks, nothing you will notice in any small business or home environments. > If you do get issues report them on the Debian BTS - I'm sure they'd get > it fixed, either by an updated package or backporting a fix to the current > version. Bugs only get fixed if they're reported to the correct place > people! > Actually this is nearly what I did already, you should have read my first message more carefully. It is about requesting more information about the issue if anybody has those, not about asking for support to resolve the issue. We know that this is caused by unixODBC already but not exactly at which point and why it helps to install an newer version. The maintainer of the Debian package asked me to provide more information so that he can understand the issue in detail to create a patch for the package in Wheezy. > In any case I don't think you'll find problems with myODBC, but obviously > test before rolling into production. > As said, the issue is with unixODBC, not the myODBC connector which is a different thing. > Wheezy is still 'testing' and shouldn't be on a production machine yet > anyway, if you need newer packages check squeeze-backports ( > http://backports.debian.org/) first. > I know. However, Wheezy in currently preparing go-live and becoming the stable stamp. We are currently doing systems integration for the Gemeinschaft PBX appliance which is going to use Wheezy as a standard in the near future. Regards, Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/52e1e1b0/attachment.html From stargray at bigmir.net Wed Jan 23 23:34:57 2013 From: stargray at bigmir.net (Anton Vojlenko) Date: Wed, 23 Jan 2013 22:34:57 +0200 Subject: [Freeswitch-users] ESL and DEBUG messages In-Reply-To: References: Message-ID: Giovanni, I can see Skype API answers only when i using fs_cli. With perl script I can't see them. 2013/1/22 Giovanni Maruzzelli > On Tue, Jan 22, 2013 at 12:51 PM, Anton Vojlenko > wrote: > > Hello, > > > > I want to check user status from the Skype client instance, but I can't > see > > Skype API answers from it. How to fix this? > > Have you tried to give: > > "fsctl loglevel 9" in addition to "console loglevel 9" ? > > > > > Script example: > > > > #!/usr/bin/perl > > use strict; > > use warnings; > > require ESL; > > > > my $host = "localhost"; > > my $port = "8021"; > > my $pass = "somepass"; > > my $con = new ESL::ESLconnection($host, $port, $pass); > > > > $con->api("console loglevel 9"); > > my $status = $con->api("skypopen sk1 GET USERSTATUS")->getBody(); > > print $status; > > $con->disconnect(); > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/3b74fa6e/attachment.html From richgration at gmail.com Thu Jan 24 00:13:51 2013 From: richgration at gmail.com (Richard Gration) Date: Wed, 23 Jan 2013 21:13:51 +0000 Subject: [Freeswitch-users] DTMF bug report Message-ID: Hi, I submitted a bug report today (FS-5047). I think I should have raised it at a slightly higher priority, it's more than minor. Is there anything I can do about that? Cheers, Rich -- Once our basic material needs are met - in my utopia, anyway - life becomes a perpetual celebration in which everyone has a talent to contribute. But we cannot levitate ourselves into that blessed condition by wishing it. We need to brace ourselves for a struggle against terrifying obstacles, both of our own making and imposed by the natural world. And the first step is to recover from the delusion that is positive thinking. -- Barbara Ehrenreich From benoit.raymond at amiconcept.com Thu Jan 24 00:41:29 2013 From: benoit.raymond at amiconcept.com (Benoit Raymond) Date: Wed, 23 Jan 2013 16:41:29 -0500 Subject: [Freeswitch-users] Retrieving hangupCause from b-leg in mod_lua In-Reply-To: References: <04fc01cdf918$4521b260$cf651720$@amiconcept.com> Message-ID: <060501cdf9b2$6d3a13b0$47ae3b10$@amiconcept.com> Hi Avi, The originate_disposition variable does indeed report the status of the b-leg regardless of the bridge to have connected or not. The bridge_hangup_cause is no longer reported (I use FS 1.3.13) but rather last_bridge_hangup_cause is reported only of the bridge completed so if the originator cancel the call, this variable won?t be there. Having said that, I executed the info application after the bridge so I could see all the variables and parameters returned. There is a DIALSTATUS variable that matches the originate_disposition but its value is a bit different than what originate_disposition returns. I haven?t finalize my script yet but looking at yours, it looks quite clean to me. Cheers, Benoit Raymond AMI Concept Inc. Business VoIP Solutions / Web Design Phone: (450) 553-1231 http://www.amiconcept.com De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Avi Marcus Envoy? : 23 janvier 2013 02:17 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Retrieving hangupCause from b-leg in mod_lua Try: cause = session:getVariable("originate_disposition") cause2=session:getVariable("bridge_hangup_cause") I'm not exactly sure what gets into which variable... But I think I needed both because sometimes "user_busy" was only in the second one. I'm not 100% sure... Here's my production error script here: https://gist.github.com/4602781 (if the user accessed via a the calling card IVR, it passes them back to the menu. Otherwise, it hangs up on them. Remove or keep as appropriate.) Please share your results if you are able to make it cleaner. -Avi Marcus BestFone On Wed, Jan 23, 2013 at 5:18 AM, Benoit Raymond wrote: Hello, Is there a way in lua to figure out why the b-leg hung up? I have an inbound call that I need to bridge so when lua is invoked, there is a session created for the inbound call and then I use session:execute(?bridge?, ?the b-leg path?) and either the call is bridged or call_timeout kicks in. I need to continue my script based on the hangup cause whether the b-leg produced a NO_ANSWER, USER_BUSY or any other condition. The session:hangupCause is of no use since my session is still active. I found the variable ?bridge_hangup_cause? on the wiki but it returns nil if I do session:getVariable(?bridge_hangup_cause?). I am using 1.3.13 Thanks much. Benoit Raymond AMI Concept Inc. Business VoIP Solutions / Web Design Phone: (450) 553-1231 http://www.amiconcept.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/05d5da8a/attachment-0001.html From avi at avimarcus.net Thu Jan 24 00:51:18 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 23 Jan 2013 23:51:18 +0200 Subject: [Freeswitch-users] Retrieving hangupCause from b-leg in mod_lua In-Reply-To: <060501cdf9b2$6d3a13b0$47ae3b10$@amiconcept.com> References: <04fc01cdf918$4521b260$cf651720$@amiconcept.com> <060501cdf9b2$6d3a13b0$47ae3b10$@amiconcept.com> Message-ID: "add emulation for asterisk DIALSTATUS magic var (r:9d98d49f) <== done during ClueCon! see ML thread" So you probably don't really want to reference that... -Avi On Wed, Jan 23, 2013 at 11:41 PM, Benoit Raymond < benoit.raymond at amiconcept.com> wrote: > Hi Avi,**** > > ** ** > > The originate_disposition variable does indeed report the status of the > b-leg regardless of the bridge to have connected or not. The > bridge_hangup_cause is no longer reported (I use FS 1.3.13) but rather > last_bridge_hangup_cause is reported only of the bridge completed so if the > originator cancel the call, this variable won?t be there.**** > > ** ** > > Having said that, I executed the info application after the bridge so I > could see all the variables and parameters returned. There is a DIALSTATUS > variable that matches the originate_disposition but its value is a bit > different than what originate_disposition returns.**** > > ** ** > > I haven?t finalize my script yet but looking at yours, it looks quite > clean to me.**** > > ** ** > > Cheers,**** > > ** ** > > *Benoit Raymond***** > > AMI Concept Inc.**** > > Business VoIP Solutions / Web Design**** > > Phone: (450) 553-1231**** > > http://www.amiconcept.com**** > > ** ** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Avi Marcus > *Envoy? :* 23 janvier 2013 02:17 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] Retrieving hangupCause from b-leg in > mod_lua**** > > ** ** > > Try:**** > > cause = session:getVariable("originate_disposition")**** > > cause2=session:getVariable("bridge_hangup_cause")**** > > ** ** > > I'm not exactly sure what gets into which variable... But I think I needed > both because sometimes "user_busy" was only in the second one. I'm not 100% > sure...**** > > ** ** > > Here's my production error script here: https://gist.github.com/4602781*** > * > > (if the user accessed via a the calling card IVR, it passes them back to > the menu. Otherwise, it hangs up on them. Remove or keep as appropriate.)* > *** > > ** ** > > Please share your results if you are able to make it cleaner.**** > > ** ** > > > **** > > -Avi Marcus > BestFone**** > > ** ** > > On Wed, Jan 23, 2013 at 5:18 AM, Benoit Raymond < > benoit.raymond at amiconcept.com> wrote:**** > > Hello,**** > > **** > > Is there a way in lua to figure out why the b-leg hung up? I have an > inbound call that I need to bridge so when lua is invoked, there is a > session created for the inbound call and then I use > session:execute(?bridge?, ?the b-leg path?) and either the call is bridged > or call_timeout kicks in. I need to continue my script based on the hangup > cause whether the b-leg produced a NO_ANSWER, USER_BUSY or any other > condition. The session:hangupCause is of no use since my session is still > active.**** > > **** > > I found the variable ?bridge_hangup_cause? on the wiki but it returns nil > if I do session:getVariable(?bridge_hangup_cause?).**** > > **** > > I am using 1.3.13**** > > **** > > Thanks much.**** > > **** > > *Benoit Raymond***** > > AMI Concept Inc.**** > > Business VoIP Solutions / Web Design**** > > Phone: (450) 553-1231**** > > http://www.amiconcept.com**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/085d5dcb/attachment.html From mail at loredo.me Thu Jan 24 01:01:31 2013 From: mail at loredo.me (Julian Pawlowski) Date: Wed, 23 Jan 2013 23:01:31 +0100 Subject: [Freeswitch-users] Debian unixODBC issues In-Reply-To: <0D1C698866F66045A6201FD0F59CAC90014716C4E8@EX.frontier.local> References: <0D1C698866F66045A6201FD0F59CAC90014716C4E8@EX.frontier.local> Message-ID: Hi Colin, On Wed, Jan 23, 2013 at 9:14 PM, Colin Mason wrote: > I had stability issues with unixODBC-2.2.14-12 on CentOS 6. Adding > Threading = 0 to my /etc/odbcinst.ini fixed the issues I was having. > Thanks a lot for the hint! We haven't tried this yet but it sounds promising after reading http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#.2Fetc.2Fodbcinst.ini_for_MySQL "Also note, (see bug below on bottom of page); If you are using the FreeSWITCH pooled resources, then you definitely need Threading = 0 for high call volume. What this does is it "disables a global mutex in the ODBC core that only allows 1 concurrent operation per process regardless of the concurrent connections." (i.e. It slows down when you have too many connections trying at one time)" Let's see if this actually helps. It could be that Threading=0 is default in unixODBC version 2.3 and that is why changing to that version resolved that issue. Br Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/9943b087/attachment.html From schoch+freeswitch.org at xwin32.com Thu Jan 24 02:19:12 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Wed, 23 Jan 2013 15:19:12 -0800 Subject: [Freeswitch-users] Caller-ID variables and another Caller-ID question Message-ID: There are several caller-ID variables defined in the wiki, and I'm trying to understand the difference: caller_id_number, caller_id_name, origination_caller_id_number, origination_caller_id_name, effective_caller_id_number, effective_caller_id_name, outbound_caller_id_number, outbound_caller_id_name, If I understand correctly, caller_id_number and caller_id_name are read-only variables that contain the caller-ID from an incoming call. I don't understand origination_caller_id_number. It says it sets it for LEG A, but I don't understand LEG A. What's the difference between origination_caller_id_number and effective_caller_id_number? As near as I can tell, the outbound_caller_id_number variable is not used by any module, but is mainly used to send information from a directory entry to an outbound dialplan. Do I have that right? In the dialplan, it seems the right way is to do this: Does this require me to set outbound_caller_id_name in *all* my directory entries? What if it's missing in one. Can I have it use a default? The next question is about the outgoing Caller-ID Name. From the SIP trace, I see the name is set both in the "From:" line and the "Remote-Party-ID:" line, in quotes (""). However, the name does not appear on the caller-ID box of the POTS line. It seems the phone company (AT&T) is getting the name from a CNAM lookup. Is this something I can change, or is it just not sent in the path between Flowroute and AT&T? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/4b6fca6e/attachment-0001.html From dvl36.ripe.nick at gmail.com Thu Jan 24 03:20:41 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Thu, 24 Jan 2013 02:20:41 +0200 Subject: [Freeswitch-users] bluetooth endpoint In-Reply-To: References: <20130120235956.69310@gmx.com> Message-ID: Sorry. Would never hear about this module. 2013/1/23 Moises Silva > On Mon, Jan 21, 2013 at 12:01 AM, Dmitry Lysenko < > dvl36.ripe.nick at gmail.com> wrote: > >> Only gsmopen exist in freeswitch at this moment for this purposes, but it >> does not support bluetooth link with cell phones. >> >> > Well, that's not quite true. I did a mod_handsfree (which is Bluetooth) > module for FreeSWITCH a year or two ago. It's a hack, pretty rough, but I > was able to receive and place calls from my iPhone with it. > > https://github.com/moises-silva/mod_handsfree > > I never submitted this to the standard FreeSWITCH tree because it's kind > of hackish, it uses D-BUS, which is a pain to write in C, so it uses some > helper python scripts that are called by FreeSWITCH for call control. Also > requires a recent version of D-BUS which CentOS does not have, I believe I > only found it available under Arch Linux. > > > *Moises Silva > **Manager, Software Engineering*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > ** > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter`| > | YouTube > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/7866d0be/attachment.html From sdevoy at bizfocused.com Thu Jan 24 04:00:03 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 23 Jan 2013 20:00:03 -0500 Subject: [Freeswitch-users] CRM and FreeSwitch Message-ID: <2eb801cdf9ce$2493af10$6dbb0d30$@bizfocused.com> A customer has asked if anyone has any experience integrating FreeSwitch into a CRM (Customer Relationship Manager)? I am a little puzzled by the request. I guess I am curious how it could be utilized. Their current CRM is browser based, so I would assume they are looking for "Call this customer" functions and maybe ROBO-Calling. Am I missing something else here? We have PC based code that will make their local Cisco phone pickup and dial a number selected from Outlook, etc. That doesn't even require FS. Maybe I am missing the big picture. I am fairly sure we are not talking about predictive dialing and that whole can of worms. Thanks for your thoughts. Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130123/bda568ff/attachment.html From avi at avimarcus.net Thu Jan 24 04:05:23 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 24 Jan 2013 03:05:23 +0200 Subject: [Freeswitch-users] CRM and FreeSwitch In-Reply-To: <2eb801cdf9ce$2493af10$6dbb0d30$@bizfocused.com> References: <2eb801cdf9ce$2493af10$6dbb0d30$@bizfocused.com> Message-ID: Other integration possibilities: - screen pop (common): When a call comes in to a user, their compute automatically searches for that phone number and opens the user's account - call logging -- keep a log of the calls in/out from the user - how long, which date -Avi On Thu, Jan 24, 2013 at 3:00 AM, Sean Devoy wrote: > A customer has asked if anyone has any experience integrating FreeSwitch > into a CRM (Customer Relationship Manager)?**** > > ** ** > > I am a little puzzled by the request. I guess I am curious how it could > be utilized. Their current CRM is browser based, so I would assume they > are looking for ?Call this customer? functions and maybe ROBO-Calling. Am > I missing something else here?**** > > ** ** > > We have PC based code that will make their local Cisco phone pickup and > dial a number selected from Outlook, etc. That doesn?t even require FS.. > Maybe I am missing the big picture. I am fairly sure we are not talking > about predictive dialing and that whole can of worms.**** > > ** ** > > Thanks for your thoughts.**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/42ec919e/attachment.html From ben at langfeld.co.uk Thu Jan 24 04:07:46 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Thu, 24 Jan 2013 01:07:46 +0000 Subject: [Freeswitch-users] CRM and FreeSwitch In-Reply-To: <2eb801cdf9ce$2493af10$6dbb0d30$@bizfocused.com> References: <2eb801cdf9ce$2493af10$6dbb0d30$@bizfocused.com> Message-ID: Ask your customer what they want to achieve? I'm not sure FreeSWITCH people have any idea what they way. Regards, Ben Langfeld On 24 January 2013 01:00, Sean Devoy wrote: > A customer has asked if anyone has any experience integrating FreeSwitch > into a CRM (Customer Relationship Manager)?**** > > ** ** > > I am a little puzzled by the request. I guess I am curious how it could > be utilized. Their current CRM is browser based, so I would assume they > are looking for ?Call this customer? functions and maybe ROBO-Calling. Am > I missing something else here?**** > > ** ** > > We have PC based code that will make their local Cisco phone pickup and > dial a number selected from Outlook, etc. That doesn?t even require FS. > Maybe I am missing the big picture. I am fairly sure we are not talking > about predictive dialing and that whole can of worms.**** > > ** ** > > Thanks for your thoughts.**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/26d9549a/attachment-0001.html From gabe at gundy.org Thu Jan 24 04:42:09 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 23 Jan 2013 18:42:09 -0700 Subject: [Freeswitch-users] CRM and FreeSwitch In-Reply-To: <2eb801cdf9ce$2493af10$6dbb0d30$@bizfocused.com> References: <2eb801cdf9ce$2493af10$6dbb0d30$@bizfocused.com> Message-ID: On Wed, Jan 23, 2013 at 6:00 PM, Sean Devoy wrote: > A customer has asked if anyone has any experience integrating FreeSwitch > into a CRM (Customer Relationship Manager)? > > I am a little puzzled by the request. No need to be puzzled, this stuff is all over the place. Here's an example: http://www.five9.com/call-center-software/crm-integration.htm It's pretty valuable if done right (BTW, I'm not suggesting five9s does is right, I've never used them). Best, Gabe From sdevoy at bizfocused.com Thu Jan 24 05:50:59 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 23 Jan 2013 21:50:59 -0500 Subject: [Freeswitch-users] CRM and FreeSwitch In-Reply-To: References: <2eb801cdf9ce$2493af10$6dbb0d30$@bizfocused.com> Message-ID: <2f3801cdf9dd$a4132860$ec397920$@bizfocused.com> Thanks for the replies and ideas Gabe and Avi. It seems Ben is wrong and FS people may have some good ideas. Thanks for the scolding Ben, always helpful when requesting new ideas. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel Gunderson Sent: Wednesday, January 23, 2013 8:42 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] CRM and FreeSwitch On Wed, Jan 23, 2013 at 6:00 PM, Sean Devoy wrote: > A customer has asked if anyone has any experience integrating > FreeSwitch into a CRM (Customer Relationship Manager)? > > I am a little puzzled by the request. No need to be puzzled, this stuff is all over the place. Here's an example: http://www.five9.com/call-center-software/crm-integration.htm It's pretty valuable if done right (BTW, I'm not suggesting five9s does is right, I've never used them). Best, Gabe _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From regis.freeswitch.org at tornad.net Thu Jan 24 09:50:12 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Thu, 24 Jan 2013 07:50:12 +0100 Subject: [Freeswitch-users] CRM and FreeSwitch In-Reply-To: <2f3801cdf9dd$a4132860$ec397920$@bizfocused.com> References: <2eb801cdf9ce$2493af10$6dbb0d30$@bizfocused.com> <2f3801cdf9dd$a4132860$ec397920$@bizfocused.com> Message-ID: Hi, We're using FS for providing homeshoring call center solution in france via http://www.web2contact.com/ (sorry, it's in french), and we're implemented it into 2 others specifique sofware for doing click2call or incoming call popup. We're switching from asterisk to freeswitch 3 years ago and it was a good idea ;o) You can contact me offlist if you want to know more. Regards, Regis 2013/1/24 Sean Devoy > Thanks for the replies and ideas Gabe and Avi. It seems Ben is wrong and > FS > people may have some good ideas. > > Thanks for the scolding Ben, always helpful when requesting new ideas. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Gabriel > Gunderson > Sent: Wednesday, January 23, 2013 8:42 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] CRM and FreeSwitch > > On Wed, Jan 23, 2013 at 6:00 PM, Sean Devoy wrote: > > A customer has asked if anyone has any experience integrating > > FreeSwitch into a CRM (Customer Relationship Manager)? > > > > I am a little puzzled by the request. > > No need to be puzzled, this stuff is all over the place. Here's an example: > http://www.five9.com/call-center-software/crm-integration.htm > > It's pretty valuable if done right (BTW, I'm not suggesting five9s does is > right, I've never used them). > > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/a5f76ce0/attachment.html From gvvsubhashkumar at gmail.com Thu Jan 24 10:00:26 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Thu, 24 Jan 2013 12:30:26 +0530 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: Moy, Did you get time to look into this issue? Thanks, Subhash. On Tue, Jan 22, 2013 at 6:19 PM, Subhash wrote: > Ok. > We did the test again,pasted the freeswitch log and uploaded the wireshark > of the both success and failed call traces.Please let me know if you have > problem in accessing the URLs . > > http://pastebin.freeswitch.org/20484 > > http://www.cloudshark.org/captures/6dc99dc2f68c -- Success > > http://www.cloudshark.org/captures/27b88b2c8ca4--- Failure > > > > > Thanks, > Subhash. > > > On Tue, Jan 22, 2013 at 12:06 AM, Michael Collins wrote: > >> That may be the case, but that doesn't actually supply the information >> you need to find out where the DTMFs are coming from. I would recommend >> doing the test that Moy suggested. At least find out where FreeSWITCH >> thinks this DTMF is coming from. >> >> -MC >> >> >> On Sun, Jan 20, 2013 at 6:43 PM, Subhash wrote: >> >>> But we don't see any problem when both endpoints are SIP. >>> >>> Thanks, >>> Subhash. >>> >>> >>> On Mon, Jan 21, 2013 at 7:12 AM, Moises Silva wrote: >>> >>>> On Fri, Jan 18, 2013 at 1:27 AM, Subhash wrote: >>>> >>>>> Yes,we are trying to bridge PSTN and VOIP call. >>>>> >>>>> We posted the log in pastebin.Please use the below URL to access the >>>>> freeswitch log. >>>>> >>>>> http://pastebin.freeswitch.org/20461 >>>>> >>>>> >>>> >>>> >>>> 1. 013-01-16 01:02:26.954165 [INFO] ftmod_sangoma_isdn_stack_rcv.c: >>>> 144 [s4c1][4:1] Received CONNECT/CONNECT ACK (suId:1 suInstId:2 >>>> spInstId:2 ces:0) >>>> 2. 2013-01-16 01:02:26.954165 [DEBUG] >>>> ftmod_sangoma_isdn_stack_hndl.c:285 [s4c1][4:1] Processing >>>> CONNECT/CONNECT ACK (suId:1 suInstId:2 spInstId:2 ces:0) >>>> 3. 2013-01-16 01:02:26.974165 [DEBUG] switch_rtp.c:3596 Correct >>>> ip/port confirmed. >>>> 4. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:1002 [s4c1][4 >>>> :1] First packet read stats: Rx queue len: 4, Rx queue size: 10 >>>> 5. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:938 [s4c1][4: >>>> 1] First packet write stats: Tx queue len: 1, Tx queue size: 5, Tx >>>> idle: 10 >>>> 6. 2013-01-16 01:02:27.694163 [DEBUG] switch_rtp.c:3797 RTP RECV >>>> DTMF 9:640 >>>> 7. 2013-01-16 01:02:27.694163 [DEBUG] switch_ivr_bridge.c:393 Send >>>> signal FreeTDM/4:1/2001 [BREAK] >>>> 8. 2013-01-16 01:02:27.694163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >>>> DTMF [9] >>>> >>>> >>>> The DTMF comes from the RTP side, not FreeTDM (as shown by the log >>>> above). You have to verify using tcpdump/wireshark if there is any real >>>> DTMF being received. >>>> >>>> I'd do exactly the same test again, but this time I would also start a >>>> wireshark trace. Then please pastebin the trace again and upload the pcap >>>> trace to cloudshark (http://www.cloudshark.org/) and paste the link >>>> here. >>>> >>>> Moy >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/0d503017/attachment-0001.html From steveayre at gmail.com Thu Jan 24 13:29:18 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 24 Jan 2013 10:29:18 +0000 Subject: [Freeswitch-users] Debian unixODBC issues In-Reply-To: References: Message-ID: On 23 January 2013 20:33, Julian Pawlowski wrote: > On Wed, Jan 23, 2013 at 9:20 PM, Steven Ayre wrote: > >> Personally I'm using FS + unixODBC + myODBC for MySQL with no stability >> issues. It should be fine. > > The issue occurs under higher loads or peaks, nothing you will notice in > any small business or home environments. > UP 0 years, 126 days, 19 hours, 37 minutes, 42 seconds, 994 milliseconds, 957 microseconds FreeSWITCH is ready 42124396 session(s) since startup If you do get issues report them on the Debian BTS - I'm sure they'd get > it fixed, either by an updated package or backporting a fix to the current > version. Bugs only get fixed if they're reported to the correct place > people! > > Actually this is nearly what I did already, you should have read my first > message more carefully. > It is about requesting more information about the issue if anybody has > those, not about asking for support to resolve the issue. We know that this > is caused by unixODBC already but not exactly at which point and why it > helps to install an newer version. > I actually intended that about whoever added the note to the Wiki since the debian maintainer obviously hadn't heard from them. If the note on the Wiki is from you, apologies. The maintainer of the Debian package asked me to provide more information > so that he can understand the issue in detail to create a patch for the > package in Wheezy. > If FS coredumps within unixODBC try: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 If a simple program can be written that demonstrates the same instability, then even better since the backtrace then might be more readable. If it's only under high load then it could be thread related so you'd need to reproduce that. In any case I don't think you'll find problems with myODBC, but obviously >> test before rolling into production. >> > > As said, the issue is with unixODBC, not the myODBC connector which is a > different thing. > I said myODBC since I haven't used unixODBC with any of the other connectors personally. Even if the crash appears to be in unixODBC it could still be in the connector used too since that code is called from unixODBC and could be some mismatch between what they're doing. -Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/29f0b3aa/attachment.html From kamil.kapturkiewicz at vccsystems.pl Thu Jan 24 12:40:45 2013 From: kamil.kapturkiewicz at vccsystems.pl (Kamil Kapturkiewicz - VCC Systems) Date: Thu, 24 Jan 2013 10:40:45 +0100 Subject: [Freeswitch-users] Problem with compilation Message-ID: <5101019D.1030702@vccsystems.pl> hi, i tried to install freeswitch on my slackware-14.0 (32bit), but there was problem with compilation: how can i fix it? i tried to install sofia-sip standalone, but there was no file named libmsg.la in destination directory. msg_parser.c: In function 'msg_hclass_offset': msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c: In function 'msg_header_add_dup': msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c: In function 'msg_header_add_dup_as': msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c: In function 'msg_header_add_make': msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c: In function 'msg_header_add_format': msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c: In function 'msg_header_insert': msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c: In function 'msg_header_remove': msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c: In function 'msg_header_remove_all': msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c: In function 'msg_header_replace': msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] msg_parser.c:2490:40: error: array subscript is above array bounds [-Werror=array-bounds] cc1: all warnings being treated as errors make[9]: *** [msg_parser.lo] Error 1 make[8]: *** [all] Error 2 Making all in sip LTCOMPILE sip_parser.lo LTCOMPILE sip_header.lo LTCOMPILE sip_util.lo LTCOMPILE sip_pref_util.lo LTCOMPILE sip_basic.lo LTCOMPILE sip_extra.lo LTCOMPILE sip_feature.lo LTCOMPILE sip_mime.lo LTCOMPILE sip_security.lo LTCOMPILE sip_event.lo LTCOMPILE sip_prack.lo LTCOMPILE sip_refer.lo LTCOMPILE sip_session.lo LTCOMPILE sip_caller_prefs.lo LTCOMPILE sip_reason.lo LTCOMPILE sip_status.lo LTCOMPILE sip_time.lo LTCOMPILE sip_tag_class.lo LTCOMPILE sip_inlined.lo LTCOMPILE sip_tag.lo LTCOMPILE sip_parser_table.lo LTCOMPILE sip_tag_ref.lo LINK libsip.la Making all in http LTCOMPILE http_parser.lo LTCOMPILE http_header.lo LTCOMPILE http_basic.lo LTCOMPILE http_extra.lo LTCOMPILE http_inlined.lo LTCOMPILE http_status.lo LTCOMPILE http_tag_class.lo LTCOMPILE http_tag.lo LTCOMPILE http_parser_table.lo LTCOMPILE http_tag_ref.lo LINK libhttp.la Making all in soa LTCOMPILE soa.lo LTCOMPILE soa_static.lo LTCOMPILE soa_tag.lo LTCOMPILE soa_tag_ref.lo LINK libsoa.la Making all in tport LTCOMPILE tport.lo LTCOMPILE tport_logging.lo LTCOMPILE tport_stub_sigcomp.lo LTCOMPILE tport_type_udp.lo LTCOMPILE tport_type_tcp.lo LTCOMPILE tport_type_sctp.lo LTCOMPILE tport_tag.lo LTCOMPILE tport_tag_ref.lo LTCOMPILE tport_type_connect.lo LTCOMPILE tport_type_tls.lo LTCOMPILE tport_tls.lo LINK libtport.la Making all in nta LTCOMPILE nta.lo LTCOMPILE nta_check.lo LTCOMPILE nta_tag.lo LTCOMPILE nta_tag_ref.lo LTCOMPILE sl_utils_print.lo LTCOMPILE sl_utils_log.lo LTCOMPILE sl_read_payload.lo LINK libnta.la Making all in nth LTCOMPILE nth_client.lo LTCOMPILE nth_server.lo LTCOMPILE nth_tag.lo LTCOMPILE nth_tag_ref.lo LINK libnth.la Making all in nea LTCOMPILE nea.lo LTCOMPILE nea_event.lo LTCOMPILE nea_server.lo LTCOMPILE nea_debug.lo LTCOMPILE nea_tag.lo LTCOMPILE nea_tag_ref.lo LINK libnea.la Making all in iptsec LTCOMPILE auth_client.lo LTCOMPILE auth_common.lo LTCOMPILE auth_digest.lo LTCOMPILE auth_module.lo LTCOMPILE auth_tag.lo LTCOMPILE auth_tag_ref.lo LTCOMPILE auth_plugin.lo LTCOMPILE auth_plugin_delayed.lo LTCOMPILE auth_module_sip.lo LTCOMPILE iptsec_debug.lo LTCOMPILE auth_module_http.lo LINK libiptsec.la Making all in nua LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la make[8]: *** No rule to make target `msg/libmsg.la', needed by `libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [/usr/src/freeswitch-1.2.5.3/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 brgds, -- Kamil Kapturkiewicz From avi at avimarcus.net Thu Jan 24 16:54:03 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 24 Jan 2013 15:54:03 +0200 Subject: [Freeswitch-users] Problem with compilation In-Reply-To: <5101019D.1030702@vccsystems.pl> References: <5101019D.1030702@vccsystems.pl> Message-ID: Hi Kamil. Please post bugs on http://jira.freeswitch.org to ensure it gets followed up on. If you need help on how to file a jira then join us in #freeswitch on irc.freenode.net Also, our wiki has some good information here: http://wiki.freeswitch.org/wiki/Reporting_Bugs Thanks, -Avi On Thu, Jan 24, 2013 at 11:40 AM, Kamil Kapturkiewicz - VCC Systems < kamil.kapturkiewicz at vccsystems.pl> wrote: > hi, > i tried to install freeswitch on my slackware-14.0 (32bit), but there > was problem with compilation: > how can i fix it? i tried to install sofia-sip standalone, but there > was no file named libmsg.la in destination directory. > > msg_parser.c: In function 'msg_hclass_offset': > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c: In function 'msg_header_add_dup': > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c: In function 'msg_header_add_dup_as': > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c: In function 'msg_header_add_make': > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c: In function 'msg_header_add_format': > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c: In function 'msg_header_insert': > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c: In function 'msg_header_remove': > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c: In function 'msg_header_remove_all': > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c: In function 'msg_header_replace': > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > msg_parser.c:2490:40: error: array subscript is above array bounds > [-Werror=array-bounds] > cc1: all warnings being treated as errors > make[9]: *** [msg_parser.lo] Error 1 > make[8]: *** [all] Error 2 > Making all in sip > LTCOMPILE sip_parser.lo > LTCOMPILE sip_header.lo > LTCOMPILE sip_util.lo > LTCOMPILE sip_pref_util.lo > LTCOMPILE sip_basic.lo > LTCOMPILE sip_extra.lo > LTCOMPILE sip_feature.lo > LTCOMPILE sip_mime.lo > LTCOMPILE sip_security.lo > LTCOMPILE sip_event.lo > LTCOMPILE sip_prack.lo > LTCOMPILE sip_refer.lo > LTCOMPILE sip_session.lo > LTCOMPILE sip_caller_prefs.lo > LTCOMPILE sip_reason.lo > LTCOMPILE sip_status.lo > LTCOMPILE sip_time.lo > LTCOMPILE sip_tag_class.lo > LTCOMPILE sip_inlined.lo > LTCOMPILE sip_tag.lo > LTCOMPILE sip_parser_table.lo > LTCOMPILE sip_tag_ref.lo > LINK libsip.la > Making all in http > LTCOMPILE http_parser.lo > LTCOMPILE http_header.lo > LTCOMPILE http_basic.lo > LTCOMPILE http_extra.lo > LTCOMPILE http_inlined.lo > LTCOMPILE http_status.lo > LTCOMPILE http_tag_class.lo > LTCOMPILE http_tag.lo > LTCOMPILE http_parser_table.lo > LTCOMPILE http_tag_ref.lo > LINK libhttp.la > Making all in soa > LTCOMPILE soa.lo > LTCOMPILE soa_static.lo > LTCOMPILE soa_tag.lo > LTCOMPILE soa_tag_ref.lo > LINK libsoa.la > Making all in tport > LTCOMPILE tport.lo > LTCOMPILE tport_logging.lo > LTCOMPILE tport_stub_sigcomp.lo > LTCOMPILE tport_type_udp.lo > LTCOMPILE tport_type_tcp.lo > LTCOMPILE tport_type_sctp.lo > LTCOMPILE tport_tag.lo > LTCOMPILE tport_tag_ref.lo > LTCOMPILE tport_type_connect.lo > LTCOMPILE tport_type_tls.lo > LTCOMPILE tport_tls.lo > LINK libtport.la > Making all in nta > LTCOMPILE nta.lo > LTCOMPILE nta_check.lo > LTCOMPILE nta_tag.lo > LTCOMPILE nta_tag_ref.lo > LTCOMPILE sl_utils_print.lo > LTCOMPILE sl_utils_log.lo > LTCOMPILE sl_read_payload.lo > LINK libnta.la > Making all in nth > LTCOMPILE nth_client.lo > LTCOMPILE nth_server.lo > LTCOMPILE nth_tag.lo > LTCOMPILE nth_tag_ref.lo > LINK libnth.la > Making all in nea > LTCOMPILE nea.lo > LTCOMPILE nea_event.lo > LTCOMPILE nea_server.lo > LTCOMPILE nea_debug.lo > LTCOMPILE nea_tag.lo > LTCOMPILE nea_tag_ref.lo > LINK libnea.la > Making all in iptsec > LTCOMPILE auth_client.lo > LTCOMPILE auth_common.lo > LTCOMPILE auth_digest.lo > LTCOMPILE auth_module.lo > LTCOMPILE auth_tag.lo > LTCOMPILE auth_tag_ref.lo > LTCOMPILE auth_plugin.lo > LTCOMPILE auth_plugin_delayed.lo > LTCOMPILE auth_module_sip.lo > LTCOMPILE iptsec_debug.lo > LTCOMPILE auth_module_http.lo > LINK libiptsec.la > Making all in nua > LTCOMPILE nua.lo > LTCOMPILE nua_common.lo > LTCOMPILE nua_stack.lo > LTCOMPILE nua_server.lo > LTCOMPILE nua_client.lo > LTCOMPILE nua_extension.lo > LTCOMPILE nua_dialog.lo > LTCOMPILE outbound.lo > LTCOMPILE nua_params.lo > LTCOMPILE nua_register.lo > LTCOMPILE nua_registrar.lo > LTCOMPILE nua_session.lo > LTCOMPILE nua_options.lo > LTCOMPILE nua_message.lo > LTCOMPILE nua_publish.lo > LTCOMPILE nua_subnotref.lo > LTCOMPILE nua_notifier.lo > LTCOMPILE nua_event_server.lo > LTCOMPILE nua_tag.lo > LTCOMPILE nua_tag_ref.lo > LINK libnua.la > make[8]: *** No rule to make target `msg/libmsg.la', needed by > `libsofia-sip-ua.la'. Stop. > make[7]: *** [all-recursive] Error 1 > Making all in packages > make[6]: *** [all-recursive] Error 1 > make[5]: *** [all] Error 2 > make[4]: *** > [/usr/src/freeswitch-1.2.5.3/libs/sofia-sip/libsofia-sip-ua/ > libsofia-sip-ua.la] > Error 2 > make[3]: *** [mod_sofia-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > brgds, > > -- > Kamil Kapturkiewicz > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/ee945c8a/attachment-0001.html From miha at softnet.si Thu Jan 24 17:09:10 2013 From: miha at softnet.si (Miha) Date: Thu, 24 Jan 2013 15:09:10 +0100 Subject: [Freeswitch-users] Lcr in easy_route, CID manipulation Message-ID: <51014086.2080909@softnet.si> Hi, I need to manipulate with CID for specific GW. IS it possible to set this in sip_profiles (it would be prefect:) )? In lcr I did use colume CID which worked ok, but is it possible to manipulate with to different kind of CID or use to regex? For example once cid is 01810000 in i must revert it to +3821810000, other time I get 003851810000 and I must convert it to +3861810000? What would be the best way for easy_route? Otherwise I will use culr, depend in source ip I will manipulte with cid number. Thanks! Miha From rhuddleston at gmail.com Thu Jan 24 17:36:34 2013 From: rhuddleston at gmail.com (Robert Huddleston) Date: Thu, 24 Jan 2013 09:36:34 -0500 Subject: [Freeswitch-users] Commit 5542d04 broke build / make current Message-ID: <510146F2.4060805@gmail.com> Commit 5542d04 for mod_voicemail broke build / make current making all mod_voicemail make[5]: Entering directory `/usr/src/freeswitch/src/mod/applications/mod_voicemail' make[6]: Entering directory `/usr/src/freeswitch/src/mod/applications/mod_voicemail' Compiling /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c... quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c -fPIC -DPIC -o .libs/mod_voicemail.o /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c: In function 'vm_fsdb_pref_greeting_get_function': /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:4981:6: error: variable 'slot' set but not used [-Werror=unused-but-set-variable] cc1: all warnings being treated as errors make[6]: *** [mod_voicemail.lo] Error 1 make[6]: Leaving directory `/usr/src/freeswitch/src/mod/applications/mod_voicemail' make[5]: *** [all] Error 1 make[5]: Leaving directory `/usr/src/freeswitch/src/mod/applications/mod_voicemail' make[4]: *** [mod_voicemail-all] Error 1 make[4]: Leaving directory `/usr/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 root at ip-10-144-18-29:/usr/src/freeswitch# From sdevoy at bizfocused.com Thu Jan 24 18:36:59 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 24 Jan 2013 10:36:59 -0500 Subject: [Freeswitch-users] Lcr in easy_route, CID manipulation In-Reply-To: <51014086.2080909@softnet.si> References: <51014086.2080909@softnet.si> Message-ID: <324b01cdfa48$a672c0a0$f35841e0$@bizfocused.com> HI Miha, If I understand correctly you get the CID from 2 different gateways in different formats and would like to handle them in a combined dialplan with uniform formats. I think when you said "other time I get 003851810000 and I must convert it to +3861810000?" You made a typo and it should have been : "other time I get 003861810000 and I must convert it to +3861810000?" If not the regex will need to be adjusted below. I would suggest you set the "context" from the gateways to different intermediate dial plans. For example context="from_gateway1" and context="from_gateway2". Then create the dial plans for each of those contexts like this: . . . For gateway2: . . . Then in your main trunk dialplan (here called "from-external") all the numbers will be in the format +38n and 7 more digits. Of course you can set more exacting regex expressions. And you should probably have a catch-all in the bottom of the "from_gatewayN" contexts to handle any CIDs that did not match our expressions. Hope that helps. Sean I Hope that helped. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Miha Sent: Thursday, January 24, 2013 9:09 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Lcr in easy_route, CID manipulation Hi, I need to manipulate with CID for specific GW. IS it possible to set this in sip_profiles (it would be prefect:) )? In lcr I did use colume CID which worked ok, but is it possible to manipulate with to different kind of CID or use to regex? For example once cid is 01810000 in i must revert it to +3821810000, other time I get 003851810000 and I must convert it to +3861810000? What would be the best way for easy_route? Otherwise I will use culr, depend in source ip I will manipulte with cid number. Thanks! Miha _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com <> Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/0dd0d159/attachment.html From bdfoster at endigotech.com Thu Jan 24 19:06:47 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 24 Jan 2013 11:06:47 -0500 Subject: [Freeswitch-users] Commit 5542d04 broke build / make current In-Reply-To: <510146F2.4060805@gmail.com> References: <510146F2.4060805@gmail.com> Message-ID: All bug reports go to JIRA: http://jira.freeswitch.org . - BDF On Thursday, January 24, 2013, Robert Huddleston wrote: > Commit 5542d04 for mod_voicemail broke build / make current > > making all mod_voicemail > make[5]: Entering directory > `/usr/src/freeswitch/src/mod/applications/mod_voicemail' > make[6]: Entering directory > `/usr/src/freeswitch/src/mod/applications/mod_voicemail' > Compiling > /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c... > quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/curl/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src > -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g > -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE > -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c > -fPIC -DPIC -o .libs/mod_voicemail.o > /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c: > In function 'vm_fsdb_pref_greeting_get_function': > > /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:4981:6: > error: variable 'slot' set but not used [-Werror=unused-but-set-variable] > cc1: all warnings being treated as errors > make[6]: *** [mod_voicemail.lo] Error 1 > make[6]: Leaving directory > `/usr/src/freeswitch/src/mod/applications/mod_voicemail' > make[5]: *** [all] Error 1 > make[5]: Leaving directory > `/usr/src/freeswitch/src/mod/applications/mod_voicemail' > make[4]: *** [mod_voicemail-all] Error 1 > make[4]: Leaving directory `/usr/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [current] Error 2 > root at ip-10-144-18-29:/usr/src/freeswitch# > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/5fb1d8a7/attachment-0001.html From andrew at cassidywebservices.co.uk Thu Jan 24 19:25:38 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 24 Jan 2013 16:25:38 +0000 Subject: [Freeswitch-users] Commit 5542d04 broke build / make current In-Reply-To: References: <510146F2.4060805@gmail.com> Message-ID: I looked, it is also on Jira. I was going to post something sarcastic about someone actually posting something on Jira but decided against it. http://jira.freeswitch.org/browse/FS-5051 On 24 January 2013 16:06, Brian Foster wrote: > > All bug reports go to JIRA: http://jira.freeswitch.org . > > - BDF > > > On Thursday, January 24, 2013, Robert Huddleston wrote: > >> Commit 5542d04 for mod_voicemail broke build / make current >> >> making all mod_voicemail >> make[5]: Entering directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> make[6]: Entering directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> Compiling >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c... >> quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/curl/include >> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >> -I/usr/src/freeswitch/libs/libteletone/src >> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g >> -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >> -DHAVE_CONFIG_H -c >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c >> -fPIC -DPIC -o .libs/mod_voicemail.o >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c: >> In function 'vm_fsdb_pref_greeting_get_function': >> >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:4981:6: >> error: variable 'slot' set but not used [-Werror=unused-but-set-variable] >> cc1: all warnings being treated as errors >> make[6]: *** [mod_voicemail.lo] Error 1 >> make[6]: Leaving directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> make[5]: *** [all] Error 1 >> make[5]: Leaving directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> make[4]: *** [mod_voicemail-all] Error 1 >> make[4]: Leaving directory `/usr/src/freeswitch/src/mod' >> make[3]: *** [all-recursive] Error 1 >> make[3]: Leaving directory `/usr/src/freeswitch/src' >> make[2]: *** [all-recursive] Error 1 >> make[2]: Leaving directory `/usr/src/freeswitch' >> make[1]: *** [all] Error 2 >> make[1]: Leaving directory `/usr/src/freeswitch' >> make: *** [current] Error 2 >> root at ip-10-144-18-29:/usr/src/freeswitch# >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/861ac81b/attachment.html From krice at freeswitch.org Thu Jan 24 19:41:03 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 24 Jan 2013 10:41:03 -0600 Subject: [Freeswitch-users] Commit 5542d04 broke build / make current In-Reply-To: Message-ID: Oh wait, not only was there a jira, its already resolved... Funny how that stuff happens..... I guess the point is no need to be a dick about it all the time.... The user had a legit complaint, and he opened a ticket, however, where he failed, was he forgot to mention the ticket in his post the mailing list... So people... Here?s a hint, don?t be a dick, and if you are asking on the mailing list, and you know you should open a ticket and you have opened a ticket already let us know the ticket number... On 1/24/13 10:25 AM, "Andrew Cassidy" wrote: > I looked, it is also on Jira. I was going to post something sarcastic about > someone actually posting something on Jira but decided against it. > > http://jira.freeswitch.org/browse/FS-5051 > > On 24 January 2013 16:06, Brian Foster wrote: >> >> All bug reports go to JIRA: http://jira.freeswitch.org . >> >> - BDF >> >> >> On Thursday, January 24, 2013, Robert Huddleston wrote: >>> Commit 5542d04 for mod_voicemail broke build / make current >>> >>> making all mod_voicemail >>> make[5]: Entering directory >>> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >>> make[6]: Entering directory >>> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >>> Compiling >>> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c... >>> quiet_libtool: compile: ?gcc -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g >>> -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >>> -DHAVE_CONFIG_H -c >>> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c >>> -fPIC -DPIC -o .libs/mod_voicemail.o >>> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c: >>> In function 'vm_fsdb_pref_greeting_get_function': >>> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:4981: >>> 6: >>> error: variable 'slot' set but not used [-Werror=unused-but-set-variable] >>> cc1: all warnings being treated as errors >>> make[6]: *** [mod_voicemail.lo] Error 1 >>> make[6]: Leaving directory >>> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >>> make[5]: *** [all] Error 1 >>> make[5]: Leaving directory >>> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >>> make[4]: *** [mod_voicemail-all] Error 1 >>> make[4]: Leaving directory `/usr/src/freeswitch/src/mod' >>> make[3]: *** [all-recursive] Error 1 >>> make[3]: Leaving directory `/usr/src/freeswitch/src' >>> make[2]: *** [all-recursive] Error 1 >>> make[2]: Leaving directory `/usr/src/freeswitch' >>> make[1]: *** [all] Error 2 >>> make[1]: Leaving directory `/usr/src/freeswitch' >>> make: *** [current] Error 2 >>> root at ip-10-144-18-29:/usr/src/freeswitch# >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/524ae101/attachment.html From bdfoster at endigotech.com Thu Jan 24 20:04:58 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 24 Jan 2013 12:04:58 -0500 Subject: [Freeswitch-users] Commit 5542d04 broke build / make current In-Reply-To: References: Message-ID: Seems like everyone posts bugs on the mailing list, so my automatic response to bugs is to direct them to JIRA. We keep getting drilled on that. Yes it probably would be different if the post had the JIRA link, but then again the idea behind bug reports going to JIRA is that all information goes there, in one spot, in a bug report. Not scattered around where it's harder for the developer who's trying to fix it. Sent from my iPhone On Jan 24, 2013, at 11:41 AM, Ken Rice wrote: > Oh wait, not only was there a jira, its already resolved... Funny how that stuff happens..... > > I guess the point is no need to be a dick about it all the time.... > > The user had a legit complaint, and he opened a ticket, however, where he failed, was he forgot to mention the ticket in his post the mailing list... > > So people... Here?s a hint, don?t be a dick, and if you are asking on the mailing list, and you know you should open a ticket and you have opened a ticket already let us know the ticket number... > > > On 1/24/13 10:25 AM, "Andrew Cassidy" wrote: > > I looked, it is also on Jira. I was going to post something sarcastic about someone actually posting something on Jira but decided against it. > > http://jira.freeswitch.org/browse/FS-5051 > > On 24 January 2013 16:06, Brian Foster wrote: > > All bug reports go to JIRA: http://jira.freeswitch.org . > > - BDF > > > On Thursday, January 24, 2013, Robert Huddleston wrote: > Commit 5542d04 for mod_voicemail broke build / make current > > making all mod_voicemail > make[5]: Entering directory > `/usr/src/freeswitch/src/mod/applications/mod_voicemail' > make[6]: Entering directory > `/usr/src/freeswitch/src/mod/applications/mod_voicemail' > Compiling > /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c... > quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/curl/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src > -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g > -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE > -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c > -fPIC -DPIC -o .libs/mod_voicemail.o > /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c: > In function 'vm_fsdb_pref_greeting_get_function': > /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:4981:6: > error: variable 'slot' set but not used [-Werror=unused-but-set-variable] > cc1: all warnings being treated as errors > make[6]: *** [mod_voicemail.lo] Error 1 > make[6]: Leaving directory > `/usr/src/freeswitch/src/mod/applications/mod_voicemail' > make[5]: *** [all] Error 1 > make[5]: Leaving directory > `/usr/src/freeswitch/src/mod/applications/mod_voicemail' > make[4]: *** [mod_voicemail-all] Error 1 > make[4]: Leaving directory `/usr/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [current] Error 2 > root at ip-10-144-18-29:/usr/src/freeswitch# > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/21073592/attachment-0001.html From rhuddleston at gmail.com Thu Jan 24 21:10:24 2013 From: rhuddleston at gmail.com (Robert-GMAIL) Date: Thu, 24 Jan 2013 13:10:24 -0500 Subject: [Freeswitch-users] Commit 5542d04 broke build / make current In-Reply-To: References: Message-ID: <3E31178D-C0B5-4F65-BE38-4C221C7E8945@gmail.com> Ya lets not be a dick to me please.. I failed to open the jira before sending the email.. So crucify me.. I went to irc and provided the jira.. So crucify me.. We are paying freeswitch for consulting.. So crucify me... Figure others might get screwed by a bad commit.. So crucify me... I swear some people on this list act like Steve Jobs or Linus T. And their crap dont stink.. Oh and new to the freeswitch community from Asterisk! Thanks the not as perfect as others man! Sent from my iPhone 5 On Jan 24, 2013, at 12:04 PM, Brian Foster wrote: > Seems like everyone posts bugs on the mailing list, so my automatic response to bugs is to direct them to JIRA. We keep getting drilled on that. Yes it probably would be different if the post had the JIRA link, but then again the idea behind bug reports going to JIRA is that all information goes there, in one spot, in a bug report. Not scattered around where it's harder for the developer who's trying to fix it. > > Sent from my iPhone > > On Jan 24, 2013, at 11:41 AM, Ken Rice wrote: > >> Oh wait, not only was there a jira, its already resolved... Funny how that stuff happens..... >> >> I guess the point is no need to be a dick about it all the time.... >> >> The user had a legit complaint, and he opened a ticket, however, where he failed, was he forgot to mention the ticket in his post the mailing list... >> >> So people... Here?s a hint, don?t be a dick, and if you are asking on the mailing list, and you know you should open a ticket and you have opened a ticket already let us know the ticket number... >> >> >> On 1/24/13 10:25 AM, "Andrew Cassidy" wrote: >> >> I looked, it is also on Jira. I was going to post something sarcastic about someone actually posting something on Jira but decided against it. >> >> http://jira.freeswitch.org/browse/FS-5051 >> >> On 24 January 2013 16:06, Brian Foster wrote: >> >> All bug reports go to JIRA: http://jira.freeswitch.org . >> >> - BDF >> >> >> On Thursday, January 24, 2013, Robert Huddleston wrote: >> Commit 5542d04 for mod_voicemail broke build / make current >> >> making all mod_voicemail >> make[5]: Entering directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> make[6]: Entering directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> Compiling >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c... >> quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/curl/include >> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >> -I/usr/src/freeswitch/libs/libteletone/src >> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g >> -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >> -DHAVE_CONFIG_H -c >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c >> -fPIC -DPIC -o .libs/mod_voicemail.o >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c: >> In function 'vm_fsdb_pref_greeting_get_function': >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:4981:6: >> error: variable 'slot' set but not used [-Werror=unused-but-set-variable] >> cc1: all warnings being treated as errors >> make[6]: *** [mod_voicemail.lo] Error 1 >> make[6]: Leaving directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> make[5]: *** [all] Error 1 >> make[5]: Leaving directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> make[4]: *** [mod_voicemail-all] Error 1 >> make[4]: Leaving directory `/usr/src/freeswitch/src/mod' >> make[3]: *** [all-recursive] Error 1 >> make[3]: Leaving directory `/usr/src/freeswitch/src' >> make[2]: *** [all-recursive] Error 1 >> make[2]: Leaving directory `/usr/src/freeswitch' >> make[1]: *** [all] Error 2 >> make[1]: Leaving directory `/usr/src/freeswitch' >> make: *** [current] Error 2 >> root at ip-10-144-18-29:/usr/src/freeswitch# >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/bee12688/attachment.html From rhuddleston at gmail.com Thu Jan 24 21:12:13 2013 From: rhuddleston at gmail.com (Robert-GMAIL) Date: Thu, 24 Jan 2013 13:12:13 -0500 Subject: [Freeswitch-users] Commit 5542d04 broke build / make current In-Reply-To: References: Message-ID: Oh and the fix before email... Thank gmail or freeswitch.org's MX.. I sent the email much earlier and the chatter on irc might have helped resolve the bad commit quicker.. Pondering further an autobuild setup might have caught this bad commit. Sent from my iPhone 5 On Jan 24, 2013, at 12:04 PM, Brian Foster wrote: > Seems like everyone posts bugs on the mailing list, so my automatic response to bugs is to direct them to JIRA. We keep getting drilled on that. Yes it probably would be different if the post had the JIRA link, but then again the idea behind bug reports going to JIRA is that all information goes there, in one spot, in a bug report. Not scattered around where it's harder for the developer who's trying to fix it. > > Sent from my iPhone > > On Jan 24, 2013, at 11:41 AM, Ken Rice wrote: > >> Oh wait, not only was there a jira, its already resolved... Funny how that stuff happens..... >> >> I guess the point is no need to be a dick about it all the time.... >> >> The user had a legit complaint, and he opened a ticket, however, where he failed, was he forgot to mention the ticket in his post the mailing list... >> >> So people... Here?s a hint, don?t be a dick, and if you are asking on the mailing list, and you know you should open a ticket and you have opened a ticket already let us know the ticket number... >> >> >> On 1/24/13 10:25 AM, "Andrew Cassidy" wrote: >> >> I looked, it is also on Jira. I was going to post something sarcastic about someone actually posting something on Jira but decided against it. >> >> http://jira.freeswitch.org/browse/FS-5051 >> >> On 24 January 2013 16:06, Brian Foster wrote: >> >> All bug reports go to JIRA: http://jira.freeswitch.org . >> >> - BDF >> >> >> On Thursday, January 24, 2013, Robert Huddleston wrote: >> Commit 5542d04 for mod_voicemail broke build / make current >> >> making all mod_voicemail >> make[5]: Entering directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> make[6]: Entering directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> Compiling >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c... >> quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/curl/include >> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >> -I/usr/src/freeswitch/libs/libteletone/src >> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g >> -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >> -DHAVE_CONFIG_H -c >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c >> -fPIC -DPIC -o .libs/mod_voicemail.o >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c: >> In function 'vm_fsdb_pref_greeting_get_function': >> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:4981:6: >> error: variable 'slot' set but not used [-Werror=unused-but-set-variable] >> cc1: all warnings being treated as errors >> make[6]: *** [mod_voicemail.lo] Error 1 >> make[6]: Leaving directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> make[5]: *** [all] Error 1 >> make[5]: Leaving directory >> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >> make[4]: *** [mod_voicemail-all] Error 1 >> make[4]: Leaving directory `/usr/src/freeswitch/src/mod' >> make[3]: *** [all-recursive] Error 1 >> make[3]: Leaving directory `/usr/src/freeswitch/src' >> make[2]: *** [all-recursive] Error 1 >> make[2]: Leaving directory `/usr/src/freeswitch' >> make[1]: *** [all] Error 2 >> make[1]: Leaving directory `/usr/src/freeswitch' >> make: *** [current] Error 2 >> root at ip-10-144-18-29:/usr/src/freeswitch# >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/9f4a877b/attachment-0001.html From alex at jajah.com Thu Jan 24 21:19:28 2013 From: alex at jajah.com (Alex Massover) Date: Thu, 24 Jan 2013 20:19:28 +0200 Subject: [Freeswitch-users] Origination of INVITE w/o SDP Message-ID: <569384504C492C4580E88B5D54DFEAEA33D99E43FF@jjex01.jajah.dublin> Hi, Is there any way of doing ESL origination of INVITE for outbound leg without SDP? FS --> INVITE w/o SDP --> B FS <-- 200 OK w SDP <--B FS --> ACK w SDP --> B (It's different from 3pcc proxy option since there's no inbound leg in my scenario). Not complex hack of mod_sofia (that doesn't break anything else) is also a good solution, if someone could point me more precisely what's the best way of doing it in the code I'd greatly appreciate. -- Best Regards, Alex Massover -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/b05078cd/attachment.html From gamar at center.com Thu Jan 24 21:25:11 2013 From: gamar at center.com (Gilbert Amar) Date: Thu, 24 Jan 2013 10:25:11 -0800 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> Message-ID: <004101cdfa60$25b1b4e0$71151ea0$@center.com> Hello, I succeeded in building mod_opal, but it crash FreeSWITCH (segmentation default) after simple test. Here are the steps on Debian "squeeze". After getting all required packages including swig Advice, check that you do not have any version of ptlib or h323plus or opal already somewhere. use find /- name "*libpt*.so" use find /- name "*libopal*.so" My FreeSWITCH is in /usr/local/src/ and I am root when I build. cd /usr/local/src/ svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_12_0/ ptlib-2.12.0 cd /usr/local/src/ptlib-2.12.0 ./configure --disable-plugins --disable-v4l2 --prefix=/opt/lib make && make install export PKG_CONFIG_PATH=/opt/lib/lib/pkgconfig cd /usr/local/src/ svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/tags/v3_12_0 opal-3.12.0 cd /usr/local/src/opal-3.12.0 ./configure --disable-plugins --prefix=/opt/lib make && make install As the install fails to copy include/ep/ cd /usr/local/src/opal-3.12.0 mkdir -p /opt/lib/include/opal/ep ; chmod 755 /opt/lib/include/opal/ep ; ( for fn in include/ep/*.h ; do /usr/bin/install -c -m 644 $fn /opt/lib/include/opal/ep ; done); Please notice that the mkdir line is one long line. Add mod_opal to your modules.conf Go to your mod_opal src dir in my case cd /usr/local/src/freeswitch/src/mod/endpoints/mod_opal Edit mod_opal.h You need to change 2 lines #include to #include then #define HAVE_T38 (OPAL_CHECK_VERSION(3,11,2) && OPAL_T38_CAPABILITY) to #define HAVE_T38 0 Now you can build mod_opal type make Go back to your FS dir and do make and make install to check all is ok. Before launch FreeSWITCH you need to set LD_LIBRARY_PATH export LD_LIBRARY_PATH=/opt/lib/lib start your FreeSWITCH Then try to call extension 9197 it should work despite the errors lines like 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not initialized! 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not initialized! 2013-01-24 03:12:25.597058 [ERR] switch_core_timer.c:117 Timer is not properly configured. In my case I have a sip phone registered to FS at extension 1000 If I call from openphone this extension the call get thru, I pick up but mod_opal isn't aware of this and when I hang up on the sip phone FreeSWITCH crash. Segmentation fault /opt/freeswitch-v1.2stable/bin/freeswitch -conf /home/center/freeswitch/conf -log /home/center/freeswitch/log -db /home/center/freeswitch/db -sounds /home/center/freeswitch/sounds -nonat -nonatmap Gilbert From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Wednesday, January 23, 2013 3:26 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] building mod_opal and v1.2stable Do you mean you built opal in /root/opal? You shouldn't build things as root, but that's not the reason for your troubles. I have to say I'm a bit puzzled, it looks like you've installed opal to multiple locations. /usr/local/freeswitch/opal looks the best match (opal/localep.h) but it's in the wrong location. /usr/local/src/freeswitch/libs/opal/include/ep/localep.h Not sure why it's installed here, opal isn't shipped under libs/ /usr/local/include/ep/localep.h /usr/include/ep/localep.h Close but the directory is wrong as it needs to be opal/localep.h, not ep/localep.h /usr/include/opal/localep.h Should be the correct filename, but you don't have a copy. It could be a change in the opal library but I find that unlikely since it'd break apps using the library like FS. More likely there's a problem with the prefixes given to opal's configure command. -Steve On 23 January 2013 10:46, Anton Rostotskiy wrote: Search results: /usr/local/freeswitch/opal/include/opal/localep.h /usr/local/src/freeswitch/libs/opal/include/ep/localep.h /usr/local/include/ep/localep.h /usr/include/ep/localep.h /root/opal/include/ep/localep.h As I said earlier, I installed ptlib and opal in /root directory. 2013/1/23 Steven Ayre It's localep.h, not local.h: /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.h:41:26: error: opal/localep.h: No such file or directory find / -name localep.h Sorry, seems my phone did an annoying autouncorrection! -Steve On 23 January 2013 08:16, Anton Rostotskiy wrote: Thanks for reply, Steven! Tried to search local.h file, but search returned nothing. That are all local.h files in my system: /usr/local/src/freeswitch/libs/spandsp/src/gsm0610_local.h /usr/local/src/freeswitch/libs/spandsp/src/t30_local.h /usr/local/src/freeswitch/libs/spandsp/src/t42_t43_local.h About -devel packages - at first tried to install ptlib, opal and -devel from yum but their versions have been too low for mod_opal. 2013/1/22 Steven Ayre Run: find / -name local.h It's failing to include opal/local.h. All the other errors are because its missing the definitions in that file. Normally I would say to instal the -devel package, but since you built from svn that doesn't apply. Perhaps it's not installed or installed to the wrong place. The find command will tell us where that file is. Steve on iPhone On 22 Jan 2013, at 16:07, Anton Rostotskiy wrote: Also tried to install mod_opal and also failed. CentOS 6.3 x86_64 ptlib and opal are the last version obtained from svn as it is said in wiki . but just a bunch of errors after make mod_opal http://pastebin.com/HHUPLCLV 2013/1/18 Steven Ayre The latest version of opal in Debian (even in Sid) isn't new enough for mod_opal. It needed various features added that are only only available in the svn version. The wiki page states you have to install ptlib and opal from svn - this is why. http://wiki.freeswitch.org/wiki/Mod_opal Of course that can then cause conflicts unless you package it yourself or install a dummy package, or problems with other libraries/apps that are built against the libopal that's already packaged. Ptlib and h323plus/opal are also notorious for having to get matching versions. If you don't pair exactly the right version of ptlib with the specific version of h323plus/opal you can get instability problems. So the official Debian ptlib package probably wouldn't be good with opal from svn... As for mod_h323, it does compile and work, at least on Wheezy and possibly Squeeze too. I've never had any luck getting mod_opal working but have got mod_h323 working. I don't actively use it though - all my traffic is currently SIP. Another option is to run Yate as a SIP->H323 proxy. It only handles the signalling, media still goes directly caller->FS. It worked quite well, although it would occasionally crash - but at least that only dropped the H323 calls and didn't take FS down with it, and Yate automatically restarted. That was on Lenny so a long time ago... Steve on iPhone On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote: Hi, I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and opal 3.10.7 on a debian Ptlib and opal get build with no error, but when trying mod_opal I get Compiling /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DP_64BIT -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, switch_frame_t**, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: error: 'class OpalCall' has no member named 'IsSwitchingT38' /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, const switch_frame_t*, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: error: 'class OpalCall' has no member named 'IsSwitchingT38' make[3]: *** [mod_opal.lo] Error 1 make[2]: *** [install] Error 1 make[1]: *** [mod_opal-install] Error 1 make: *** [mod_opal-install] Error 2 Has anyone succeded in building mod_opal ? Should I go with mod_h323 instead ? Thanks for your help _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- ? ?????????, ????? ????????? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- ? ?????????, ????? ????????? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- ? ?????????, ????? ????????? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/a3a22f22/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.log.bz2 Type: application/octet-stream Size: 19330 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/a3a22f22/attachment-0001.obj From freeswitch-list at puzzled.xs4all.nl Thu Jan 24 22:02:08 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Thu, 24 Jan 2013 20:02:08 +0100 Subject: [Freeswitch-users] Commit 5542d04 broke build / make current In-Reply-To: <3E31178D-C0B5-4F65-BE38-4C221C7E8945@gmail.com> References: <3E31178D-C0B5-4F65-BE38-4C221C7E8945@gmail.com> Message-ID: <51018530.2080401@puzzled.xs4all.nl> On 01/24/2013 07:10 PM, Robert-GMAIL wrote: [snip knickers in a twist stuff :-)] > Oh and new to the freeswitch community from Asterisk! Welcome! Hope you enjoy setting up your FreeSWITCH box(es) and sit back, relax and just see things fly. > Thanks > the not as perfect as others man! Nobody is. Positive spin: everybody reading this thread will probably add the Jira next time they want to post about a (possible) bug. I know I will. And that makes it easier for the developers to fix stuff which benefits us all. Thanks for supporting the project by paying for Consulting. Regards, Patrick From freeswitch-list at puzzled.xs4all.nl Thu Jan 24 22:04:01 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Thu, 24 Jan 2013 20:04:01 +0100 Subject: [Freeswitch-users] Commit 5542d04 broke build / make current In-Reply-To: References: Message-ID: <510185A1.9090205@puzzled.xs4all.nl> On 01/24/2013 07:12 PM, Robert-GMAIL wrote: > Oh and the fix before email... Thank gmail or freeswitch.org > 's MX.. I sent the email much earlier and the > chatter on irc might have helped resolve the bad commit quicker.. I have seen slow delivery to the mailinglist too. Sometimes servers get bogged down. Nothing new. Just file it under "things happen" :) > Pondering further an autobuild setup might have caught this bad commit. Afaik there is a Jenkins CI VM doing continuous builds. Regards, Patrick From mkdutchman at gmail.com Thu Jan 24 22:31:57 2013 From: mkdutchman at gmail.com (Melvin King) Date: Thu, 24 Jan 2013 14:31:57 -0500 Subject: [Freeswitch-users] Voicemail, control CID playback Message-ID: Hi all, Is there a way to allow the user to play or not play the caller id associated with a particular voice mail message? the skip-info key does skip the time and date but not the cid info, in other words, I have to listen to each caller's number every time I listen to the message. Is there a better way? I'm using Freeswitch 32 bit version 1.2.5.3 Mel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/1f39bd83/attachment.html From jaykris at gmail.com Thu Jan 24 22:50:02 2013 From: jaykris at gmail.com (JP) Date: Thu, 24 Jan 2013 11:50:02 -0800 Subject: [Freeswitch-users] Passing credentials when invoking "curl" Message-ID: I am trying to use curl from a lua script to get data from a webservice and need to pass user/password information. Can someone help me with this? Here is what I am doing, which does not seem to work. api = freeswitch.API() agent_status_url = "-u userid:password http://mywebservice.com:8080/status/agent/test" response = api:execute("curl", agent_status_url) freeswitch.consoleLog("INFO", response) Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/3a248f8c/attachment.html From gamar at center.com Thu Jan 24 23:04:57 2013 From: gamar at center.com (Gilbert Amar) Date: Thu, 24 Jan 2013 12:04:57 -0800 Subject: [Freeswitch-users] Passing credentials when invoking "curl" In-Reply-To: References: Message-ID: <000601cdfa6e$159a6940$40cf3bc0$@center.com> Hi If it is a basic authentication, you can try http://userid:password at mywebservice.com:8080/status/agent/test not sure it works try first a direct from command line : curl "http://userid:password at mywebservice.com:8080/status/agent/test" Gilbert From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of JP Sent: Thursday, January 24, 2013 11:50 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Passing credentials when invoking "curl" I am trying to use curl from a lua script to get data from a webservice and need to pass user/password information. Can someone help me with this? Here is what I am doing, which does not seem to work. api = freeswitch.API() agent_status_url = "-u userid:password http://mywebservice.com:8080/status/agent/test" response = api:execute("curl", agent_status_url) freeswitch.consoleLog("INFO", response) Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/aed40bc1/attachment.html From burakozturk.ce at gmail.com Thu Jan 24 19:03:49 2013 From: burakozturk.ce at gmail.com (=?ISO-8859-1?B?QnVyYWsg1np0/HJr?=) Date: Thu, 24 Jan 2013 18:03:49 +0200 Subject: [Freeswitch-users] Freeswitch TLS certificate control Message-ID: Hi, I m a new FS user. I am trying to use FS with TLS as a SIP server. My client application is CSipSimple. According to my tests, FS do not need to verify itself to client or do not verify clients' certificates. Maybe there are some problems at the client side but i thought it is not. But I m also open for new, fresh ideas about FS-CSipSimple configuretions. I read wiki and some mail-group posts. There is a paramater "tls-verify-policy". If I didn't understand wrong, I must set it with "all" value in sofia.conf.xml. I m not sure that if I set it true or wrong. Also not sure about the syntax of sofia.conf.xml file. Can anyone tell me what is the correct syntax for sofia config file for adding a new parameter. Or if I am wrong, please, tell me the correct solution. Thank you for all. Best regards, Burak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/75b3f38c/attachment.html From bdfoster at endigotech.com Fri Jan 25 00:01:29 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 24 Jan 2013 16:01:29 -0500 Subject: [Freeswitch-users] Commit 5542d04 broke build / make current In-Reply-To: <3E31178D-C0B5-4F65-BE38-4C221C7E8945@gmail.com> References: <3E31178D-C0B5-4F65-BE38-4C221C7E8945@gmail.com> Message-ID: <9C45037D-15EB-4C1A-B99E-72F13E1BF518@endigotech.com> The idea was simple, bug reports go to JIRA. I usually have a canned response but I'm mobile today, so I kept it short. That doesn't mean I'm being a dick, I'm not. Just simply encouraging folks to check out JIRA when they need to file a bug, to make it easier on those who need to fix it and quicker response in your perspective. As far as the image, I use that a lot. It's funny, not a slam on you or anyone else. So please be nice :) Sent from my iPhone On Jan 24, 2013, at 1:10 PM, Robert-GMAIL wrote: > Ya lets not be a dick to me please.. I failed to open the jira before sending the email.. So crucify me.. > > I went to irc and provided the jira.. So crucify me.. > > We are paying freeswitch for consulting.. So crucify me... > > Figure others might get screwed by a bad commit.. So crucify me... > > I swear some people on this list act like Steve Jobs or Linus T. And their crap dont stink.. > > Oh and new to the freeswitch community from Asterisk! > > Thanks > the not as perfect as others man! > > > Sent from my iPhone 5 > > On Jan 24, 2013, at 12:04 PM, Brian Foster wrote: > >> Seems like everyone posts bugs on the mailing list, so my automatic response to bugs is to direct them to JIRA. We keep getting drilled on that. Yes it probably would be different if the post had the JIRA link, but then again the idea behind bug reports going to JIRA is that all information goes there, in one spot, in a bug report. Not scattered around where it's harder for the developer who's trying to fix it. >> >> Sent from my iPhone >> >> On Jan 24, 2013, at 11:41 AM, Ken Rice wrote: >> >>> Oh wait, not only was there a jira, its already resolved... Funny how that stuff happens..... >>> >>> I guess the point is no need to be a dick about it all the time.... >>> >>> The user had a legit complaint, and he opened a ticket, however, where he failed, was he forgot to mention the ticket in his post the mailing list... >>> >>> So people... Here?s a hint, don?t be a dick, and if you are asking on the mailing list, and you know you should open a ticket and you have opened a ticket already let us know the ticket number... >>> >>> >>> On 1/24/13 10:25 AM, "Andrew Cassidy" wrote: >>> >>> I looked, it is also on Jira. I was going to post something sarcastic about someone actually posting something on Jira but decided against it. >>> >>> http://jira.freeswitch.org/browse/FS-5051 >>> >>> On 24 January 2013 16:06, Brian Foster wrote: >>> >>> All bug reports go to JIRA: http://jira.freeswitch.org . >>> >>> - BDF >>> >>> >>> On Thursday, January 24, 2013, Robert Huddleston wrote: >>> Commit 5542d04 for mod_voicemail broke build / make current >>> >>> making all mod_voicemail >>> make[5]: Entering directory >>> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >>> make[6]: Entering directory >>> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >>> Compiling >>> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c... >>> quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> -I/usr/src/freeswitch/libs/stfu -fPIC -Werror -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g >>> -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >>> -DHAVE_CONFIG_H -c >>> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c >>> -fPIC -DPIC -o .libs/mod_voicemail.o >>> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c: >>> In function 'vm_fsdb_pref_greeting_get_function': >>> /usr/src/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:4981:6: >>> error: variable 'slot' set but not used [-Werror=unused-but-set-variable] >>> cc1: all warnings being treated as errors >>> make[6]: *** [mod_voicemail.lo] Error 1 >>> make[6]: Leaving directory >>> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >>> make[5]: *** [all] Error 1 >>> make[5]: Leaving directory >>> `/usr/src/freeswitch/src/mod/applications/mod_voicemail' >>> make[4]: *** [mod_voicemail-all] Error 1 >>> make[4]: Leaving directory `/usr/src/freeswitch/src/mod' >>> make[3]: *** [all-recursive] Error 1 >>> make[3]: Leaving directory `/usr/src/freeswitch/src' >>> make[2]: *** [all-recursive] Error 1 >>> make[2]: Leaving directory `/usr/src/freeswitch' >>> make[1]: *** [all] Error 2 >>> make[1]: Leaving directory `/usr/src/freeswitch' >>> make: *** [current] Error 2 >>> root at ip-10-144-18-29:/usr/src/freeswitch# >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> irc.freenode.net #freeswitch >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/1b5f0db1/attachment-0001.html From jaykris at gmail.com Fri Jan 25 00:45:00 2013 From: jaykris at gmail.com (JP) Date: Thu, 24 Jan 2013 13:45:00 -0800 Subject: [Freeswitch-users] Passing credentials when invoking "curl" In-Reply-To: <000601cdfa6e$159a6940$40cf3bc0$@center.com> References: <000601cdfa6e$159a6940$40cf3bc0$@center.com> Message-ID: Thanks Gilbert. That worked for me. -JP On Thu, Jan 24, 2013 at 12:04 PM, Gilbert Amar wrote: > Hi **** > > If it is a basic authentication, you can try **** > > http://userid:password at mywebservice.com:8080/status/agent/test**** > > ** ** > > not sure it works try first a direct from command line :**** > > ** ** > > curl "http://userid:password at mywebservice.com:8080/status/agent/test" **** > > ** ** > > Gilbert**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *JP > *Sent:* Thursday, January 24, 2013 11:50 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Passing credentials when invoking "curl"**** > > ** ** > > I am trying to use curl from a lua script to get data from a webservice > and need to pass user/password information. Can someone help me with this? > Here is what I am doing, which does not seem to work.**** > > ** ** > > api = freeswitch.API()**** > > agent_status_url = "-u userid:password > http://mywebservice.com:8080/status/agent/test"**** > > response = api:execute("curl", agent_status_url)**** > > freeswitch.consoleLog("INFO", response)**** > > ** ** > > Thanks**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/d216fd92/attachment.html From lists at kavun.ch Fri Jan 25 02:22:57 2013 From: lists at kavun.ch (Emrah) Date: Thu, 24 Jan 2013 18:22:57 -0500 Subject: [Freeswitch-users] Cannot save Voicemail message In-Reply-To: References: <3263505F-2AC8-4BB5-ABEB-44B706AD80C5@kavun.ch> <6770CA93-7D95-404A-809A-732A68C60771@kavun.ch> Message-ID: <32167726-6AA5-4FFF-B44E-BD8BF58DB645@kavun.ch> Yes I can reproduce this on the latest git head. Can someone confirm? My VMs are recorded in MP3 format. Thanks On Jan 21, 2013, at 1:29 PM, Michael Collins wrote: > Indeed, no information at all. See if you can reproduce this on latest git HEAD. If you can, please open a Jira so that the devs can track this issue. Also, if this was working on a previous version of FreeSWITCH it would be very helpful to find out which commit broke it. > > -MC > > On Sat, Jan 19, 2013 at 4:31 PM, Emrah wrote: > The console logs are not very helpful? > > http://pastebin.freeswitch.org/20467 > > Thanks for any help. > > E > On Jan 18, 2013, at 6:24 PM, Michael Collins wrote: > > > Can you supply a bit more context? Show us more log lines. > > -MC > > > > On Fri, Jan 18, 2013 at 2:18 PM, Emrah wrote: > > Hi all, > > > > I've noticed a strange problem lately, I can't press 3 to save a VM. > > > > I use mod_shout, my VMs are in MP3 format. > > > > Log abstract: > > > > 2013-01-18 17:11:09.191700 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 16000hz 1 channels 20ms > > 2013-01-18 17:11:10.191704 [DEBUG] switch_rtp.c:3827 RTP RECV DTMF 3:1120 > > 2013-01-18 17:11:10.191704 [DEBUG] switch_ivr_play_say.c:1682 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-save_recording.wav > > 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] > > 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-you_have.wav] (en:en) > > > > freeswitch at internal> version > > FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z) > > > > Deleting works fine. > > Any idea? > > > > Thanks and all the best, > > Emrah > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From schoch+freeswitch.org at xwin32.com Fri Jan 25 03:10:55 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 24 Jan 2013 16:10:55 -0800 Subject: [Freeswitch-users] Nested Condition In-Reply-To: <50F96329.9050007@gmail.com> References: <50F96329.9050007@gmail.com> Message-ID: On Fri, Jan 18, 2013 at 6:58 AM, Mimiko wrote: > I try: > > > expression="^${pref1}([5-8]\d\d)$" require-nested="true"> > > > > > expression="^${pref2}([6-8]\d\d)$" require-nested="true"> > > > > > > How about: I think that does pretty much the same thing, but I'm not sure of your exact requirements. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/c897bb2f/attachment.html From schoch+freeswitch.org at xwin32.com Fri Jan 25 03:44:31 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 24 Jan 2013 16:44:31 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> Message-ID: I've tried a few more things, but it's still not there. Here's the latest log from the Windows machine: http://pastebin.freeswitch.org/20500 I've been told that I need to enable T.38, because the txfax module doesn't do that, but I can't figure out how. Any suggestions? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/b157cdea/attachment.html From schoch+freeswitch.org at xwin32.com Fri Jan 25 04:23:33 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 24 Jan 2013 17:23:33 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> Message-ID: On Wed, Jan 23, 2013 at 9:19 AM, Michael Collins wrote: > I find it interesting that the ZRTP engine threw a warning. Do you see any > ZRTP messages on the Linux log where the fax actually worked? No, it just worked, with none of those messages. The Linux log is here: http://pastebin.freeswitch.org/20501 The Windows log is here: http://pastebin.freeswitch.org/20500 As usual, everything is working just fine on Linux. As soon as it touches Microsoft, it breaks. On Linux, setting "total-modems" to 2 produced /dev/FS0 and /dev/FS1. On Windows, I don't see any additional COM ports. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/38f56038/attachment-0001.html From jaykris at gmail.com Fri Jan 25 04:34:13 2013 From: jaykris at gmail.com (JP) Date: Thu, 24 Jan 2013 17:34:13 -0800 Subject: [Freeswitch-users] Executing multiple applications in parallel Message-ID: I want to do the following... 1. Play a long music file from a dialplan. 2. While the music is playing, I want to launch a Lua script to do some data dip. 3. Once the data is fetched and Lua exits, I want to interrupt the music and continue in the dialplan How do I do this. Any sample code or pointers would be greatly appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130124/1eb313d5/attachment.html From pashdown at xmission.com Fri Jan 25 02:57:09 2013 From: pashdown at xmission.com (Pete Ashdown) Date: Thu, 24 Jan 2013 16:57:09 -0700 Subject: [Freeswitch-users] Two Freeswitches - Invalid Gateway Message-ID: <5101CA55.9040006@xmission.com> Hello, I'm trying to do something that appears outwardly simple, but is causing me grief. I asked the question on the IRC channel, but I'm still waiting for the ticket to be answered. I followed the instructions for connecting two Freeswitch boxes here: http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes The result is that I get an "Invalid Gateway" when trying to dial extensions on either box, like it is trying to go through the SIP external profile. Any thoughts as to what I'm doing wrong, or what is missing from the above configuration? Thanks in advance. From freeswitch at nullinfinity.org Fri Jan 25 06:05:13 2013 From: freeswitch at nullinfinity.org (Johan Brannlund) Date: Thu, 24 Jan 2013 22:05:13 -0500 Subject: [Freeswitch-users] PSTN via Google Talk Message-ID: <20130124220513.6b1b1ba0@mars> Hello. I've been trying to get PSTN dialing through Google Talk to work, but the call setup just hangs for 15 seconds and then terminates with 2013-01-19 23:07:44.237479 [DEBUG] mod_dingaling.c:4425 hungup dingaling/gtalk/+ 18008727245 at voice.google.com 2013-01-19 23:07:44.237479 [DEBUG] mod_dingaling.c:839 Terminate called from lin e 4426 state=CS_INIT 2013-01-19 23:07:44.237479 [DEBUG] switch_channel.c:2994 (dingaling/gtalk/+18008 727245 at voice.google.com) Callstate Change DOWN -> HANGUP 2013-01-19 23:07:44.237479 [NOTICE] mod_dingaling.c:864 Hangup dingaling/gtalk/+18008727245 at voice.google.com [CS_INIT] [NORMAL_CLEARING] Full log: http://pastebin.freeswitch.org/20502 My jingle profile: http://pastebin.freeswitch.org/20503 Some other data points: * The gmail account in question has google talk set up (I can make PSTN calls from within gmail), but not google voice (not available in Canada). * The gmail account is not signed in via any web browsers. However, it is the same account that's setup on my Android phone. * I've noticed that some documentation says to use address at gmail.com/talk in the jingle profile, while some says address at gmail.com/gtalk. I'm not sure if that's significant. * I tried a few different STUN servers, but all with the same result. Any help would be much appreciated. From ssoni at lifesize.com Fri Jan 25 09:37:28 2013 From: ssoni at lifesize.com (Sanjay Soni) Date: Fri, 25 Jan 2013 00:37:28 -0600 Subject: [Freeswitch-users] PSTN via Google Talk In-Reply-To: <20130124220513.6b1b1ba0@mars> References: <20130124220513.6b1b1ba0@mars> Message-ID: I see these lines in your log 2013-01-19 23:07:28.257512 [DEBUG] mod_dingaling.c:1697 Don't have an audio codec. 2013-01-19 23:07:28.257512 [DEBUG] mod_dingaling.c:1712 Don't have my video codec yet here's one 2013-01-19 23:07:28.257512 [DEBUG] mod_dingaling.c:1740 Send Describe [(null)@0] 2013-01-19 23:07:29.117479 [DEBUG] mod_dingaling.c:4105 using Existing session for 5371162578 2013-01-19 23:07:29.117479 [DEBUG] mod_dingaling.c:1697 Don't have an audio codec. Have you specified audio codec in autoload_config/dingaling.conf.xml as ? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Johan Brannlund Sent: 25 January 2013 08:35 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] PSTN via Google Talk Hello. I've been trying to get PSTN dialing through Google Talk to work, but the call setup just hangs for 15 seconds and then terminates with 2013-01-19 23:07:44.237479 [DEBUG] mod_dingaling.c:4425 hungup dingaling/gtalk/+ 18008727245 at voice.google.com 2013-01-19 23:07:44.237479 [DEBUG] mod_dingaling.c:839 Terminate called from lin e 4426 state=CS_INIT 2013-01-19 23:07:44.237479 [DEBUG] switch_channel.c:2994 (dingaling/gtalk/+18008 727245 at voice.google.com) Callstate Change DOWN -> HANGUP 2013-01-19 23:07:44.237479 [NOTICE] mod_dingaling.c:864 Hangup dingaling/gtalk/+18008727245 at voice.google.com [CS_INIT] [NORMAL_CLEARING] Full log: http://pastebin.freeswitch.org/20502 My jingle profile: http://pastebin.freeswitch.org/20503 Some other data points: * The gmail account in question has google talk set up (I can make PSTN calls from within gmail), but not google voice (not available in Canada). * The gmail account is not signed in via any web browsers. However, it is the same account that's setup on my Android phone. * I've noticed that some documentation says to use address at gmail.com/talk in the jingle profile, while some says address at gmail.com/gtalk. I'm not sure if that's significant. * I tried a few different STUN servers, but all with the same result. Any help would be much appreciated. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From Sirish.MasurMohan at oa.com.au Fri Jan 25 09:41:33 2013 From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan) Date: Fri, 25 Jan 2013 17:41:33 +1100 Subject: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error Message-ID: <965759A53E43FE439E43565A7715E5F0591C085DB2@oa-exchange1.oa.com.au> Hello All, I think I am missing something obvious here, but I am having trouble trying to make calls between two SNOM phones. I have an open ACL for registration and I keep getting BEARERCAPABILITY_NOTAUTH error when I try to make calls. Would appreciate any inputs on how to debug this! Logs are at : http://pastebin.freeswitch.org/20504 With regards, Sirish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/d609d131/attachment.html From miha at softnet.si Fri Jan 25 10:27:27 2013 From: miha at softnet.si (Miha) Date: Fri, 25 Jan 2013 08:27:27 +0100 Subject: [Freeswitch-users] Lcr in easy_route, CID manipulation In-Reply-To: <324b01cdfa48$a672c0a0$f35841e0$@bizfocused.com> References: <51014086.2080909@softnet.si> <324b01cdfa48$a672c0a0$f35841e0$@bizfocused.com> Message-ID: <510233DF.1010707@softnet.si> Hi Sean, thank you for your replay. I need to manipulate with two different cid on one GW but I should made it so scapable that if I would add another GW and I will be getting on this GW different CID than on first one, that I could use different regex:) I as have written my dialplan with xml_curl I would do it there, with php this should be easy:) Thanks for you info! Miha Dne 1/24/2013 4:36 PM, pis(e Sean Devoy: > > HI Miha, > > If I understand correctly you get the CID from 2 different gateways in > different formats and would like to handle them in a combined dialplan > with uniform formats. > > I think when you said "other time I get 003851810000 and I must > convert it to +3861810000?" You made a typo and it should have been : > "other time I get 003861810000 and I must convert it to +3861810000?" > If not the regex will need to be adjusted below. > > I would suggest you set the "context" from the gateways to different > intermediate dial plans. For example context="from_gateway1" and > context="from_gateway2". > > Then create the dial plans for each of those contexts like this: > > > > > > > > data="982$1 XML from-external"/> > > > > > > . . . > > > > For gateway2: > > > > > > > > > > > > > > . . . > > > > Then in your main trunk dialplan (here called "from-external") all the > numbers will be in the format +38n and 7 more digits. > > Of course you can set more exacting regex expressions. And you should > probably have a catch-all in the bottom of the "from_gatewayN" > contexts to handle any CIDs that did not match our expressions. > > Hope that helps. > > Sean > > I Hope that helped. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Miha > Sent: Thursday, January 24, 2013 9:09 AM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Lcr in easy_route, CID manipulation > > Hi, > > I need to manipulate with CID for specific GW. IS it possible to set > this in sip_profiles (it would be prefect:) )? > > In lcr I did use colume CID which worked ok, but is it possible to > manipulate with to different kind of CID or use to regex? For example > once cid is 01810000 in i must revert it to +3821810000, other time I > get 003851810000 and I must convert it to +3861810000? > > What would be the best way for easy_route? > > Otherwise I will use culr, depend in source ip I will manipulte with > cid number. > > Thanks! > > Miha > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/5b6e7fb4/attachment-0001.html From itsusama at gmail.com Fri Jan 25 10:29:15 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Fri, 25 Jan 2013 12:29:15 +0500 Subject: [Freeswitch-users] ESL Perl Fork Message-ID: Hi Guys, So I'm kinda stuck with this non trivial issue few might ever encounter; Please have a look at the script here http://pastebin.freeswitch.org/20508 Issue is that when the child returns it closes the esl connection and I don't receive any further events on the script, a hack would be to check the connection state each time I loop in the main thread and connect if state = 0. Ideally I would like to prevent the child to mess up the connection. Thnaks. -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/38c48feb/attachment.html From avi at avimarcus.net Fri Jan 25 12:22:31 2013 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 25 Jan 2013 11:22:31 +0200 Subject: [Freeswitch-users] Executing multiple applications in parallel In-Reply-To: References: Message-ID: ESL gives you more control, especially to execute things in parallel. If you want to do it within FS, that's a little more complicated, since most things are sequential. However, if you use the async APIs, then I think it could work: 1) sched_broadcast -- schedule the playback of your file on the UUID of the channel, for +0, meaning now. 2) run your lua script 3) Then stop (break) the playback on that uuid: break 4) Continue with your execution. Please wikify your results and let us know how that works out! -Avi On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: > I want to do the following... > > 1. Play a long music file from a dialplan. > 2. While the music is playing, I want to launch a Lua script to do some > data dip. > 3. Once the data is fetched and Lua exits, I want to interrupt the music > and continue in the dialplan > > How do I do this. Any sample code or pointers would be greatly appreciated. > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/d631b3e4/attachment.html From alessio at asistar.it Fri Jan 25 14:21:05 2013 From: alessio at asistar.it (Alessio) Date: Fri, 25 Jan 2013 12:21:05 +0100 Subject: [Freeswitch-users] Patton FXO configuration on Freeswitch Message-ID: <51026AA1.9040806@asistar.it> Hi all! I am trying to configure a Patton FXO Gateway in my Freeswitch, but I don't know how and where i have to write configuration file. Anyone can help me? Thanks, Alessio From matt at inveroak.com Fri Jan 25 17:27:13 2013 From: matt at inveroak.com (Matt Broad) Date: Fri, 25 Jan 2013 14:27:13 +0000 Subject: [Freeswitch-users] Remove inband DTMF on record_session In-Reply-To: References: Message-ID: Hi, sorry to interject into your message, but I was looking to do something similar to Tamas. I have the need to prevent the DTMF from being recorded when using record_session. I am using RFC2833 (i think :) ), is this at all possible in the current Freeswitch build? Matt On 21 January 2013 07:25, Tamas Jalsovszky wrote: > Hello Moy, > > Inserting silence instead of the tone would be fine for me. I think the >= > 50ms delay in recording is not an issue. I guess, removing from "live" > streams would be harder and would introduce the delay you mentioned. > So you have a plan to add such a functionality in the near future? That > would be amazing :) > > Regards, > Jalsot > > On Mon, Jan 21, 2013 at 2:48 AM, Moises Silva wrote: > >> On Sun, Jan 20, 2013 at 4:12 PM, Tamas Jalsovszky wrote: >> >>> Hello, >>> >>> Is there a way to to remove inbad DTMF from the recorded wav file >>> (conditionaly)? >>> If not, how could be that done the easyiest way and a way it could get >>> into FS git? >>> >>> As FS has a great DTMF detector and it has probably all the stuff doing >>> such a feature, I think it would be much better than using an external tool >>> for removing DTMF from finished wav file - and it would consume more I/O >>> probably. >>> >>> >> Removing it would not be that hard (ie, inserting silence instead). >> Removing it without leaving a trace any DTMF was ever there sounds a tad >> more difficult. If you're ok with inserting a silence instead of the tone, >> extending the record_session app to detect and remove DTMF is probably what >> I'd do. >> >> It is also possible to create a new app that monitors the audio stream >> and silence the stream when DTMF is detected, but since the detection takes >> a few milliseconds (~50ms), there would be a bit of "bleeding" of DTMF in >> the signal, unless the new app also delays the signal by 50ms and then has >> a way to strip the DTMF completely. >> >> *Moises Silva >> **Manager, Software Engineering*** >> >> msilva at sangoma.com >> >> Sangoma Technologies >> >> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >> >> >> t. +1 800 388 2475 (N. America) >> >> t. +1 905 474 1990 x128 >> >> f. +1 905 474 9223 >> >> >> >> ** >> >> Products >> | Solutions >> | Events >> | Contact >> | Wiki >> | Facebook >> | Twitter`| >> | YouTube >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/ac068413/attachment-0001.html From alessio at asistar.it Fri Jan 25 18:55:34 2013 From: alessio at asistar.it (Alessio) Date: Fri, 25 Jan 2013 16:55:34 +0100 Subject: [Freeswitch-users] Patton FXO configuration on Freeswitch - more details Message-ID: <5102AAF6.2070201@asistar.it> Hi all! We're trying to configure a Patton FXO in our Freeswitch. We have an internal PBX - Freeswitch 1.3.6 installed from svn master with (demo) setup on IP 10.0.1.190 - Patton 4112 with simple config on IP 10.0.1.10 Find more info below. So any help would be great! We set it up that the Patton registers with the PBX and we get the error: --------------------- 2013-01-25 16:44:39.433873 [WARNING] sofia_reg.c:2469 Can't find user [fxo0_3_0 at 10.0.1.190] You must define a domain called '10.0.1.190' in your directory and add a user with the id="fxo0_3_0" attribute and you must configure your device to use the proper domain in it's authentication credentials. --------------------- We defined a gateway in conf/sip_profiles/external/fxo.xml: --------------------- --> --------------------- And the relevant parts of the Patton config: --------------------- # define auth authentication-service authenticate username fxo0_3_0 password gtatFAePDQFL0btl3cbsaA== encrypted # patton registers location-service LOCATION_FXO_3_0 domain 1 10.0.1.190 5060 identity-group group_fxo0_3_0 authentication outbound authenticate 1 authentication-service authenticate username fxo0_3_0 identity fxo0_3_0 authentication outbound authenticate 1 authentication-service authenticate username fxo0_3_0 registration outbound registrar 10.0.1.190 5060 lifetime 180 register auto retry-timeout on-system-error 10 retry-timeout on-client-error 10 retry-timeout on-server-error 10 call outbound use profile tone-set IT use profile voip default use profile sip default preferred-transport-protocol tcp traffic-class local-default invite-transaction-timeout 32 non-invite-transaction-timeout 32 penalty-box call inbound use profile tone-set IT use profile voip default use profile sip default context sip-gateway ASISTAR_3_0 interface fxo0_3_0 bind interface eth0 context router port 5060 context sip-gateway ASISTAR_3_0 bind location-service LOCATION_FXO_3_0 no shutdown --------------------- Output from 'sofia status' --------------------- external::10.0.1.10 gateway sip:fxo0_3_0 at 10.0.1.10 NOREG Thanks, Alessio From krice at freeswitch.org Fri Jan 25 19:14:44 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 25 Jan 2013 10:14:44 -0600 Subject: [Freeswitch-users] Friday Free For All Message-ID: Hey Guys, Its Friday, you know what that means... Stop by the bridge and see whats going on... Officially it doesn?t start til like 3P eastern, but its a free for all so we can start any time... Asking yourself how can I get in on this? Well sip:888 at conference.freeswitch.org will get you there or use one of our handy PSTN access numbers USA +1-919-386-9900 Spain +34-91-290-12-71 UK +44-3300-100-295 Germany +49-228-9293-9009 Australia +61-7-3188-7519 Israel +972-2-372-0394 Canada +1-438-800-0531 France +33-975-181-606 Netherlands +31-858-880-387 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/31f3f5cf/attachment.html From steveu at coppice.org Fri Jan 25 19:15:15 2013 From: steveu at coppice.org (Steve Underwood) Date: Sat, 26 Jan 2013 00:15:15 +0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: References: <20130119191943.ca09ab6d@mail.tritonwest.net> Message-ID: <5102AF93.2000201@coppice.org> On 01/25/2013 09:23 AM, Steven Schoch wrote: > On Wed, Jan 23, 2013 at 9:19 AM, Michael Collins > wrote: > > I find it interesting that the ZRTP engine threw a warning. Do you > see any ZRTP messages on the Linux log where the fax actually worked? > > > No, it just worked, with none of those messages. The Linux log is here: > http://pastebin.freeswitch.org/20501 > The Windows log is here: > http://pastebin.freeswitch.org/20500 > > As usual, everything is working just fine on Linux. As soon as it > touches Microsoft, it breaks. > > On Linux, setting "total-modems" to 2 produced /dev/FS0 and /dev/FS1. > On Windows, I don't see any additional COM ports. Your previous messages indicate that you are use txfax, so why are you configuring FAX modems? If you really want FAX modems, because you are trying to work with HylaFAX, Windows FAX, or some other FAX server which talks to FAX modems, the setup is a little different between Linux and Windows. Windows doesn't have a direct equivalent to a Linux PTY, so you need to install com0com, which gives you some back-to-back virtual com ports. The details for that are in the wiki. Steve From steveu at coppice.org Fri Jan 25 19:29:10 2013 From: steveu at coppice.org (Steve Underwood) Date: Sat, 26 Jan 2013 00:29:10 +0800 Subject: [Freeswitch-users] Friday Free For All In-Reply-To: References: Message-ID: <5102B2D6.7020601@coppice.org> On 01/26/2013 12:14 AM, Ken Rice wrote: > Friday Free For All Hey Guys, > > Its Friday, you know what that means... Yes. Its POETS day! > > Stop by the bridge and see whats going on... > > Officially it doesn?t start til like 3P eastern, but its a free for > all so we can start any time... > > Asking yourself how can I get in on this? > > Well sip:888 at conference.freeswitch.org will get you there or use one > of our handy PSTN access numbers > > USA +1-919-386-9900 > Spain +34-91-290-12-71 > UK +44-3300-100-295 > Germany +49-228-9293-9009 > Australia +61-7-3188-7519 > Israel +972-2-372-0394 > Canada +1-438-800-0531 > France +33-975-181-606 > Netherlands +31-858-880-387 Steve From steveayre at gmail.com Fri Jan 25 20:12:46 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 25 Jan 2013 17:12:46 +0000 Subject: [Freeswitch-users] Remove inband DTMF on record_session In-Reply-To: References: Message-ID: If you're using RFC2833 then that's not in the audio itself so won't be recorded. It is possible though that the caller is ending DTMF both in and out of band (which is fine as long as you only try to detect one or the other but not both). -Steve On 25 January 2013 14:27, Matt Broad wrote: > Hi, > > sorry to interject into your message, but I was looking to do something > similar to Tamas. > > I have the need to prevent the DTMF from being recorded when using > record_session. I am using RFC2833 (i think :) ), is this at all possible > in the current Freeswitch build? > > Matt > > > On 21 January 2013 07:25, Tamas Jalsovszky wrote: > >> Hello Moy, >> >> Inserting silence instead of the tone would be fine for me. I think the >> >= 50ms delay in recording is not an issue. I guess, removing from "live" >> streams would be harder and would introduce the delay you mentioned. >> So you have a plan to add such a functionality in the near future? That >> would be amazing :) >> >> Regards, >> Jalsot >> >> On Mon, Jan 21, 2013 at 2:48 AM, Moises Silva wrote: >> >>> On Sun, Jan 20, 2013 at 4:12 PM, Tamas Jalsovszky wrote: >>> >>>> Hello, >>>> >>>> Is there a way to to remove inbad DTMF from the recorded wav file >>>> (conditionaly)? >>>> If not, how could be that done the easyiest way and a way it could get >>>> into FS git? >>>> >>>> As FS has a great DTMF detector and it has probably all the stuff doing >>>> such a feature, I think it would be much better than using an external tool >>>> for removing DTMF from finished wav file - and it would consume more I/O >>>> probably. >>>> >>>> >>> Removing it would not be that hard (ie, inserting silence instead). >>> Removing it without leaving a trace any DTMF was ever there sounds a tad >>> more difficult. If you're ok with inserting a silence instead of the tone, >>> extending the record_session app to detect and remove DTMF is probably what >>> I'd do. >>> >>> It is also possible to create a new app that monitors the audio stream >>> and silence the stream when DTMF is detected, but since the detection takes >>> a few milliseconds (~50ms), there would be a bit of "bleeding" of DTMF in >>> the signal, unless the new app also delays the signal by 50ms and then has >>> a way to strip the DTMF completely. >>> >>> *Moises Silva >>> **Manager, Software Engineering*** >>> >>> msilva at sangoma.com >>> >>> Sangoma Technologies >>> >>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >>> >>> >>> t. +1 800 388 2475 (N. America) >>> >>> t. +1 905 474 1990 x128 >>> >>> f. +1 905 474 9223 >>> >>> >>> >>> ** >>> >>> Products >>> | Solutions >>> | Events >>> | Contact >>> | Wiki >>> | Facebook >>> | Twitter`| >>> | YouTube >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/ccbcccb7/attachment-0001.html From schoch+freeswitch.org at xwin32.com Fri Jan 25 21:28:09 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 25 Jan 2013 10:28:09 -0800 Subject: [Freeswitch-users] FAX problems In-Reply-To: <5102AF93.2000201@coppice.org> References: <20130119191943.ca09ab6d@mail.tritonwest.net> <5102AF93.2000201@coppice.org> Message-ID: On Fri, Jan 25, 2013 at 8:15 AM, Steve Underwood wrote: > Your previous messages indicate that you are use txfax, so why are you > configuring FAX modems? > I'm using txfax as a test. The goal is to allow a Windows 7 user to send FAXes directly from Word, QuickBooks, etc. > Windows doesn't have a direct equivalent to a Linux PTY, so you need to > install com0com, which gives you some back-to-back virtual com ports. > Now I understand. I can do that. > The details for that are in the wiki. > Can you be more specific? http://wiki.freeswitch.org/wiki/Special:Search?search=com0com "Your search - *com0com* - did not match any documents." -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/eb4dea43/attachment.html From sipman.voip at gmail.com Fri Jan 25 18:47:18 2013 From: sipman.voip at gmail.com (Marc Godbout) Date: Fri, 25 Jan 2013 10:47:18 -0500 Subject: [Freeswitch-users] Voicemail and forwarding MWI In-Reply-To: <20130114165933.29c9a361@mars> References: <20130114165933.29c9a361@mars> Message-ID: Hi, It would be nice to have a way to configure a profile were we could put some rules to forward some SIP messages, something like relaying. That could be used for the subscribe, notify, etc. Regards, -- Jimmy Check out www.voicemeup.com On Mon, Jan 14, 2013 at 4:59 PM, Johan Brannlund < freeswitch at nullinfinity.org> wrote: > > Hello. I recently installed Freeswitch for use as a home PBX. I was > planning on using my VoIP provider's voicemail system, but I'm not > sure if that's possible. In particular, I don't know how to get MWI > working in such a configuration. > > Is there a way for Freeswitch to forward the SIP NOTIFY MWI messages > from my provider to a certain extension? > > In other words, I am trying to do exactly the same thing that Gabriel > Burca asked about a year ago (see > > http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078703.html > ). > > Any help would be much appreciated. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/076e031b/attachment.html From msc at freeswitch.org Fri Jan 25 22:17:36 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 11:17:36 -0800 Subject: [Freeswitch-users] Patton FXO configuration on Freeswitch - more details In-Reply-To: <5102AAF6.2070201@asistar.it> References: <5102AAF6.2070201@asistar.it> Message-ID: Alessio, That "error" is just explaining to you that the caller does not have an entry in the XML directory, therefore there's no way to perform authentication. You'll need to create a user with the ID mentioned in the error message and then set the password in there as well as in the Patton. (I've never done a Patton set up but I've done several other analog gateway devices.) Hope this helps. -Michael On Fri, Jan 25, 2013 at 7:55 AM, Alessio wrote: > Hi all! > > We're trying to configure a Patton FXO in our Freeswitch. > > We have an internal PBX > - Freeswitch 1.3.6 installed from svn master with (demo) setup on IP > 10.0.1.190 > - Patton 4112 with simple config on IP 10.0.1.10 > > Find more info below. > So any help would be great! > > We set it up that the Patton registers with the PBX and we get the > error: > > --------------------- > 2013-01-25 16:44:39.433873 [WARNING] sofia_reg.c:2469 Can't find user > [fxo0_3_0 at 10.0.1.190] > You must define a domain called '10.0.1.190' in your directory and add a > user with the id="fxo0_3_0" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > --------------------- > > We defined a gateway in conf/sip_profiles/external/fxo.xml: > --------------------- > > --> > > > > > > --------------------- > > And the relevant parts of the Patton config: > --------------------- > # define auth > authentication-service authenticate > username fxo0_3_0 password gtatFAePDQFL0btl3cbsaA== encrypted > > # patton registers > location-service LOCATION_FXO_3_0 > domain 1 10.0.1.190 5060 > identity-group group_fxo0_3_0 > authentication outbound > authenticate 1 authentication-service authenticate username fxo0_3_0 > identity fxo0_3_0 > authentication outbound > authenticate 1 authentication-service authenticate username fxo0_3_0 > > registration outbound > registrar 10.0.1.190 5060 > lifetime 180 > register auto > retry-timeout on-system-error 10 > retry-timeout on-client-error 10 > retry-timeout on-server-error 10 > > call outbound > use profile tone-set IT > use profile voip default > use profile sip default > preferred-transport-protocol tcp > traffic-class local-default > invite-transaction-timeout 32 > non-invite-transaction-timeout 32 > penalty-box > > call inbound > use profile tone-set IT > use profile voip default > use profile sip default > > context sip-gateway ASISTAR_3_0 > interface fxo0_3_0 > bind interface eth0 context router port 5060 > > context sip-gateway ASISTAR_3_0 > bind location-service LOCATION_FXO_3_0 > no shutdown > > --------------------- > Output from 'sofia status' > --------------------- > > external::10.0.1.10 gateway sip:fxo0_3_0 at 10.0.1.10 NOREG > > > > Thanks, > Alessio > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/7b38f4b5/attachment.html From msc at freeswitch.org Fri Jan 25 23:07:52 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 12:07:52 -0800 Subject: [Freeswitch-users] Voicemail and forwarding MWI In-Reply-To: References: <20130114165933.29c9a361@mars> Message-ID: True, but now you're venturing far into the SIP proxy side of things. At some point we have to accept that a B2BUA and a proxy are different tools for different jobs. Yeah, you could hammer a nail home with a screwdriver because it's right there but at some point you might as well pick up the hammer... -MC On Fri, Jan 25, 2013 at 7:47 AM, Marc Godbout wrote: > Hi, > > It would be nice to have a way to configure a profile were we could put > some rules to forward some SIP messages, something like relaying. That > could be used for the subscribe, notify, etc. > > Regards, > > -- > > Jimmy > > Check out www.voicemeup.com > > On Mon, Jan 14, 2013 at 4:59 PM, Johan Brannlund < > freeswitch at nullinfinity.org> wrote: > >> >> Hello. I recently installed Freeswitch for use as a home PBX. I was >> planning on using my VoIP provider's voicemail system, but I'm not >> sure if that's possible. In particular, I don't know how to get MWI >> working in such a configuration. >> >> Is there a way for Freeswitch to forward the SIP NOTIFY MWI messages >> from my provider to a certain extension? >> >> In other words, I am trying to do exactly the same thing that Gabriel >> Burca asked about a year ago (see >> >> http://lists.freeswitch.org/pipermail/freeswitch-users/2011-December/078703.html >> ). >> >> Any help would be much appreciated. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/aee6a520/attachment-0001.html From msc at freeswitch.org Fri Jan 25 23:09:27 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 12:09:27 -0800 Subject: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error In-Reply-To: <965759A53E43FE439E43565A7715E5F0591C085DB2@oa-exchange1.oa.com.au> References: <965759A53E43FE439E43565A7715E5F0591C085DB2@oa-exchange1.oa.com.au> Message-ID: I confess that I haven't seen this specific error before. Could you get the SIP trace of this call and put it on pastebin as well? That might give you the information you need to crack the case. -MC On Thu, Jan 24, 2013 at 10:41 PM, Sirish Masur Mohan < Sirish.MasurMohan at oa.com.au> wrote: > Hello All,**** > > ** ** > > I think I am missing something obvious here, but I am having trouble > trying to make calls between two SNOM phones. I have an open ACL for > registration and I keep getting BEARERCAPABILITY_NOTAUTH error when I try > to make calls. **** > > ** ** > > Would appreciate any inputs on how to debug this! Logs are at : > http://pastebin.freeswitch.org/20504**** > > ** ** > > With regards,**** > > Sirish**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/b7d6eb98/attachment.html From msc at freeswitch.org Fri Jan 25 23:12:45 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 12:12:45 -0800 Subject: [Freeswitch-users] Cannot save Voicemail message In-Reply-To: <32167726-6AA5-4FFF-B44E-BD8BF58DB645@kavun.ch> References: <3263505F-2AC8-4BB5-ABEB-44B706AD80C5@kavun.ch> <6770CA93-7D95-404A-809A-732A68C60771@kavun.ch> <32167726-6AA5-4FFF-B44E-BD8BF58DB645@kavun.ch> Message-ID: Sounds worthy of a Jira. Go ahead and open one up and put this information in there. Feel free to reply to this thread with the jira link so that it's easy for people to follow up. -MC On Thu, Jan 24, 2013 at 3:22 PM, Emrah wrote: > Yes I can reproduce this on the latest git head. > > Can someone confirm? > > My VMs are recorded in MP3 format. > > Thanks > On Jan 21, 2013, at 1:29 PM, Michael Collins wrote: > > > Indeed, no information at all. See if you can reproduce this on latest > git HEAD. If you can, please open a Jira so that the devs can track this > issue. Also, if this was working on a previous version of FreeSWITCH it > would be very helpful to find out which commit broke it. > > > > -MC > > > > On Sat, Jan 19, 2013 at 4:31 PM, Emrah wrote: > > The console logs are not very helpful? > > > > http://pastebin.freeswitch.org/20467 > > > > Thanks for any help. > > > > E > > On Jan 18, 2013, at 6:24 PM, Michael Collins wrote: > > > > > Can you supply a bit more context? Show us more log lines. > > > -MC > > > > > > On Fri, Jan 18, 2013 at 2:18 PM, Emrah wrote: > > > Hi all, > > > > > > I've noticed a strange problem lately, I can't press 3 to save a VM. > > > > > > I use mod_shout, my VMs are in MP3 format. > > > > > > Log abstract: > > > > > > 2013-01-18 17:11:09.191700 [DEBUG] switch_ivr_play_say.c:1309 Codec > Activated L16 at 16000hz 1 channels 20ms > > > 2013-01-18 17:11:10.191704 [DEBUG] switch_rtp.c:3827 RTP RECV DTMF > 3:1120 > > > 2013-01-18 17:11:10.191704 [DEBUG] switch_ivr_play_say.c:1682 done > playing file > /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-save_recording.wav > > > 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:67 No > language specified - Using [en] > > > 2013-01-18 17:11:10.431720 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[voicemail/vm-you_have.wav] (en:en) > > > > > > freeswitch at internal> version > > > FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git > e04eab7 2012-12-29 00:17:59Z) > > > > > > Deleting works fine. > > > Any idea? > > > > > > Thanks and all the best, > > > Emrah > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > -- > > > Michael S Collins > > > Twitter: @mercutioviz > > > http://www.FreeSWITCH.org > > > http://www.ClueCon.com > > > http://www.OSTAG.org > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/407a31c4/attachment.html From matt at inveroak.com Fri Jan 25 23:14:57 2013 From: matt at inveroak.com (Matt Broad) Date: Fri, 25 Jan 2013 20:14:57 +0000 Subject: [Freeswitch-users] Remove inband DTMF on record_session In-Reply-To: References: Message-ID: Thanks for the response Steve, is there a way of checking the dtmf type? I have tried looking at the Sofia profile but cannot see any details regarding dtmf. I assume that this is determined by my sip provider? Thanks Matt On Friday, 25 January 2013, Steven Ayre wrote: > If you're using RFC2833 then that's not in the audio itself so won't be > recorded. It is possible though that the caller is ending DTMF both in and > out of band (which is fine as long as you only try to detect one or the > other but not both). > > -Steve > > > > On 25 January 2013 14:27, Matt Broad wrote: > > Hi, > > sorry to interject into your message, but I was looking to do something > similar to Tamas. > > I have the need to prevent the DTMF from being recorded when using > record_session. I am using RFC2833 (i think :) ), is this at all possible > in the current Freeswitch build? > > Matt > > > On 21 January 2013 07:25, Tamas Jalsovszky wrote: > > Hello Moy, > > Inserting silence instead of the tone would be fine for me. I think the >= > 50ms delay in recording is not an issue. I guess, removing from "live" > streams would be harder and would introduce the delay you mentioned. > So you have a plan to add such a functionality in the near future? That > would be amazing :) > > Regards, > Jalsot > > On Mon, Jan 21, 2013 at 2:48 AM, Moises Silva wrote: > > On Sun, Jan 20, 2013 at 4:12 PM, Tamas Jalsovszky wrote: > > Hello, > > Is there a way to to remove inbad DTMF from the recorded wav file > (conditionaly)? > If not, how could be that done the easyiest way and a way it could get > into FS git? > > As FS has a great DTMF detector and it has probably all the stuff doing > such a feature, I think it would be much better than using an external tool > for removing DTMF from finished wav file - and it would consume more I/O > probably. > > > Removing it would not be that hard (ie, inserting silence instead). > Removing it without leaving a trace any DTMF was ever there sounds a tad > more difficult. If you're ok with inserting a silence instead of the tone, > extending the record_session app to detect and remove DTMF is probably what > I'd do. > > It is also possible to create a new app that monitors the audio stream and > silence the stream when DTMF is detected, but since the detection takes a > few milliseconds (~50ms), there would be a bit of "bleeding" of DTMF in the > signal, unless the new app also delays the signal by 50ms and then has a > way to strip the DTMF completely. > > *Moises Silva > **Manager, Software Engineering*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > -- Thanks Matt This email and any attachments to it are confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of InverOak Limited. If you are not the intended recipient of this email, you must neither take any action based upon its contents, nor copy or show it to anyone. Please contact the sender if you believe you have received this email in error. This email including any attachments cannot be guaranteed to be 100% secure or error-free as information could be intercepted, corrupted, lost, destroyed, out-dated, or containing viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. InverOak Limited is a company registered in England & Wales under company number 04529594, whose registered address is Old Barn house, 2 Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/3945b13c/attachment-0001.html From msc at freeswitch.org Fri Jan 25 23:21:22 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 12:21:22 -0800 Subject: [Freeswitch-users] Caller-ID variables and another Caller-ID question In-Reply-To: References: Message-ID: On Wed, Jan 23, 2013 at 3:19 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > There are several caller-ID variables defined in the wiki, and I'm trying > to understand the difference: > > caller_id_number, caller_id_name, > origination_caller_id_number, origination_caller_id_name, > effective_caller_id_number, effective_caller_id_name, > outbound_caller_id_number, outbound_caller_id_name, > > If I understand correctly, caller_id_number and caller_id_name are > read-only variables that contain the caller-ID from an incoming call. > Correct. > > I don't understand origination_caller_id_number. It says it sets it for > LEG A, but I don't understand LEG A. > This one is for when you don't already have a leg, i.e. when you're doing an originate at the fs_cli. Example: freeswitch at internal>originate {origination_caller_id_number=18005551212}sofia/gateway/mygw/14083425400 1002 > > What's the difference between origination_caller_id_number and > effective_caller_id_number? > Use effective_caller_id_number in a bridge: > > As near as I can tell, the outbound_caller_id_number variable is not used > by any module, but is mainly used to send information from a directory > entry to an outbound dialplan. Do I have that right? > That is correct. > In the dialplan, it seems the right way is to do this: > > data="effective_caller_id_name=${outbound_caller_id_name}"/> > > Does this require me to set outbound_caller_id_name in *all* my directory > entries? What if it's missing in one. Can I have it use a default? > You can set this in vars.xml. Look for the line with "0000000000" and you'll be able to figure out the rest. > The next question is about the outgoing Caller-ID Name. From the SIP > trace, I see the name is set both in the "From:" line and the > "Remote-Party-ID:" line, in quotes (""). However, the name does not appear > on the caller-ID box of the POTS line. It seems the phone company (AT&T) > is getting the name from a CNAM lookup. Is this something I can change, or > is it just not sent in the path between Flowroute and AT&T? > If the phone company is doing a CNAM lookup then there's not really anything you can do, unless they have a reason that they're doing a CNAM lookup instead of using the caller_id_name information. I'd ask Flowroute for feedback on that one. -MC > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/83c8a0af/attachment.html From msc at freeswitch.org Fri Jan 25 23:28:29 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 12:28:29 -0800 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: <004101cdfa60$25b1b4e0$71151ea0$@center.com> References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> <004101cdfa60$25b1b4e0$71151ea0$@center.com> Message-ID: Gilbert, Thank you for collecting all of this information. You did quite a bit of work and that will help the developers figure out what's going on. Would you mind creating a ticket at jira.freeswitch.org and adding all this information? That will help the developers keep track of the issue and avoid any unnecessary delays. Thanks! -MC On Thu, Jan 24, 2013 at 10:25 AM, Gilbert Amar wrote: > Hello,**** > > ** ** > > I succeeded in building mod_opal, but it crash FreeSWITCH (segmentation > default) after simple test.**** > > ** ** > > Here are the steps on Debian "squeeze".**** > > After getting all required packages including swig**** > > ** ** > > Advice, check that you do not have any version of ptlib or h323plus or > opal already somewhere.**** > > use find /- name "*libpt*.so"**** > > use find /- name "*libopal*.so"**** > > ** ** > > ** ** > > ** ** > > My FreeSWITCH is in /usr/local/src/ and I am root when I build.**** > > ** ** > > cd /usr/local/src/**** > > svn co > https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_12_0/ptlib-2.12.0 > **** > > ** ** > > cd /usr/local/src/ptlib-2.12.0**** > > ./configure --disable-plugins --disable-v4l2 --prefix=/opt/lib**** > > make && make install**** > > ** ** > > export PKG_CONFIG_PATH=/opt/lib/lib/pkgconfig **** > > ** ** > > ** ** > > cd /usr/local/src/**** > > svn co > https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/tags/v3_12_0opal-3.12.0 > **** > > ** ** > > cd /usr/local/src/opal-3.12.0**** > > ./configure --disable-plugins --prefix=/opt/lib**** > > make && make install**** > > ** ** > > ** ** > > As the install fails to copy include/ep/**** > > ** ** > > cd /usr/local/src/opal-3.12.0**** > > mkdir -p /opt/lib/include/opal/ep ; chmod 755 /opt/lib/include/opal/ep ; ( > for fn in include/ep/*.h ; do /usr/bin/install -c -m 644 $fn > /opt/lib/include/opal/ep ; done);**** > > ** ** > > Please notice that the mkdir line is one long line.**** > > ** ** > > Add mod_opal to your modules.conf**** > > ** ** > > Go to your mod_opal src dir in my case **** > > cd /usr/local/src/freeswitch/src/mod/endpoints/mod_opal**** > > ** ** > > Edit mod_opal.h**** > > You need to change 2 lines**** > > ** ** > > #include **** > > to**** > > #include **** > > ** ** > > then **** > > #define HAVE_T38 (OPAL_CHECK_VERSION(3,11,2) && OPAL_T38_CAPABILITY)**** > > to**** > > #define HAVE_T38 0**** > > ** ** > > ** ** > > Now you can build mod_opal type make **** > > Go back to your FS dir and do make and make install to check all is ok.*** > * > > Before launch FreeSWITCH you need to set LD_LIBRARY_PATH**** > > ** ** > > export LD_LIBRARY_PATH=/opt/lib/lib**** > > start your FreeSWITCH **** > > ** ** > > Then try to call extension 9197 it should work despite the errors lines > like **** > > 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not > initialized!**** > > 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not > initialized!**** > > 2013-01-24 03:12:25.597058 [ERR] switch_core_timer.c:117 Timer is not > properly configured.**** > > ** ** > > In my case I have a sip phone registered to FS at extension 1000**** > > If I call from openphone this extension**** > > the call get thru, I pick up but mod_opal isn't aware of this and when I > hang up on the sip phone FreeSWITCH crash.**** > > ** ** > > ** ** > > Segmentation fault /opt/freeswitch-v1.2stable/bin/freeswitch -conf > /home/center/freeswitch/conf -log /home/center/freeswitch/log -db > /home/center/freeswitch/db -sounds /home/center/freeswitch/sounds -nonat > -nonatmap**** > > ** ** > > ** ** > > Gilbert**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* Wednesday, January 23, 2013 3:26 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] building mod_opal and v1.2stable**** > > ** ** > > Do you mean you built opal in /root/opal? You shouldn't build things as > root, but that's not the reason for your troubles.**** > > ** ** > > I have to say I'm a bit puzzled, it looks like you've installed opal to > multiple locations.**** > > ** ** > > /usr/local/freeswitch/opal looks the best match (opal/localep.h) but it's > in the wrong location.**** > > ** ** > > /usr/local/src/freeswitch/libs/opal/include/ep/localep.h**** > > Not sure why it's installed here, opal isn't shipped under libs/**** > > ** ** > > /usr/local/include/ep/localep.h**** > > /usr/include/ep/localep.h**** > > Close but the directory is wrong as it needs to be opal/localep.h, not > ep/localep.h**** > > ** ** > > /usr/include/opal/localep.h**** > > Should be the correct filename, but you don't have a copy.**** > > ** ** > > It could be a change in the opal library but I find that unlikely since > it'd break apps using the library like FS. More likely there's a problem > with the prefixes given to opal's configure command.**** > > ** ** > > -Steve**** > > ** ** > > ** ** > > ** ** > > On 23 January 2013 10:46, Anton Rostotskiy wrote: > **** > > Search results:**** > > ** ** > > /usr/local/freeswitch/opal/include/opal/localep.h**** > > /usr/local/src/freeswitch/libs/opal/include/ep/localep.h**** > > /usr/local/include/ep/localep.h**** > > /usr/include/ep/localep.h**** > > /root/opal/include/ep/localep.h**** > > ** ** > > As I said earlier, I installed ptlib and opal in /root directory.**** > > ** ** > > 2013/1/23 Steven Ayre **** > > It's localep.h, not local.h:**** > > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.h:41:26: > error: opal/localep.h: No such file or directory**** > > ** ** > > find / -name localep.h**** > > ** ** > > Sorry, seems my phone did an annoying autouncorrection!**** > > ** ** > > -Steve**** > > ** ** > > ** ** > > ** ** > > On 23 January 2013 08:16, Anton Rostotskiy wrote: > **** > > Thanks for reply, Steven!**** > > Tried to search local.h file, but search returned nothing. That are all > local.h files in my system:**** > > ** ** > > /usr/local/src/freeswitch/libs/spandsp/src/gsm0610_local.h**** > > /usr/local/src/freeswitch/libs/spandsp/src/t30_local.h**** > > /usr/local/src/freeswitch/libs/spandsp/src/t42_t43_local.h**** > > ** ** > > About -devel packages - at first tried to install ptlib, opal and > -devel from yum but their versions have been too low for mod_opal.**** > > ** ** > > 2013/1/22 Steven Ayre **** > > Run:**** > > find / -name local.h**** > > ** ** > > It's failing to include opal/local.h. All the other errors are because its > missing the definitions in that file.**** > > ** ** > > Normally I would say to instal the -devel package, but since you built > from svn that doesn't apply. Perhaps it's not installed or installed to the > wrong place. The find command will tell us where that file is.**** > > > Steve on iPhone**** > > > On 22 Jan 2013, at 16:07, Anton Rostotskiy wrote: > **** > > Also tried to install mod_opal and also failed.**** > > CentOS 6.3 x86_64**** > > ptlib and opal are the last version obtained from svn as it is said in > wiki .**** > > but just a bunch of errors after make mod_opal > http://pastebin.com/HHUPLCLV**** > > ** ** > > 2013/1/18 Steven Ayre **** > > The latest version of opal in Debian (even in Sid) isn't new enough for > mod_opal. It needed various features added that are only only available in > the svn version.**** > > ** ** > > The wiki page states you have to install ptlib and opal from svn - this is > why.**** > > http://wiki.freeswitch.org/wiki/Mod_opal > > Of course that can then cause conflicts unless you package it yourself or > install a dummy package, or problems with other libraries/apps that are > built against the libopal that's already packaged.**** > > ** ** > > Ptlib and h323plus/opal are also notorious for having to get matching > versions. If you don't pair exactly the right version of ptlib with the > specific version of h323plus/opal you can get instability problems. So the > official Debian ptlib package probably wouldn't be good with opal from > svn...**** > > ** ** > > As for mod_h323, it does compile and work, at least on Wheezy and possibly > Squeeze too. I've never had any luck getting mod_opal working but have got > mod_h323 working. I don't actively use it though - all my traffic is > currently SIP.**** > > ** ** > > Another option is to run Yate as a SIP->H323 proxy. It only handles the > signalling, media still goes directly caller->FS. It worked quite well, > although it would occasionally crash - but at least that only dropped the > H323 calls and didn't take FS down with it, and Yate automatically > restarted. That was on Lenny so a long time ago...**** > > > Steve on iPhone**** > > ** ** > > ** ** > > > On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote:**** > > Hi, > > I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and opal > 3.10.7 on a debian > Ptlib and opal get build with no error, but when trying mod_opal I get > > Compiling > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... > quiet_libtool: compile: g++ -g -ggdb > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src > -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE > -DP_64BIT > -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions > -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL > -DHAVE_CONFIG_H -c > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC > -DPIC -o .libs/mod_opal.o > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In > member > function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, > switch_frame_t**, switch_io_flag_t)': > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: > error: 'class OpalCall' has no member named 'IsSwitchingT38' > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In > member > function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, > const switch_frame_t*, switch_io_flag_t)': > /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: > error: 'class OpalCall' has no member named 'IsSwitchingT38' > make[3]: *** [mod_opal.lo] Error 1 > make[2]: *** [install] Error 1 > make[1]: *** [mod_opal-install] Error 1 > make: *** [mod_opal-install] Error 2 > > > Has anyone succeded in building mod_opal ? > > Should I go with mod_h323 instead ? > > Thanks for your help > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > ? ?????????, ????? ????????? **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > ? ?????????, ????? ????????? **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > ? ?????????, ????? ????????? **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/8453a86b/attachment-0001.html From msc at freeswitch.org Fri Jan 25 23:29:19 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 12:29:19 -0800 Subject: [Freeswitch-users] Passing credentials when invoking "curl" In-Reply-To: References: <000601cdfa6e$159a6940$40cf3bc0$@center.com> Message-ID: Awesome! Now it's time to pay the wiki tax! :) -MC On Thu, Jan 24, 2013 at 1:45 PM, JP wrote: > Thanks Gilbert. That worked for me. > -JP > > On Thu, Jan 24, 2013 at 12:04 PM, Gilbert Amar wrote: > >> Hi **** >> >> If it is a basic authentication, you can try **** >> >> http://userid:password at mywebservice.com:8080/status/agent/test**** >> >> ** ** >> >> not sure it works try first a direct from command line :**** >> >> ** ** >> >> curl "http://userid:password at mywebservice.com:8080/status/agent/test" *** >> * >> >> ** ** >> >> Gilbert**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *JP >> *Sent:* Thursday, January 24, 2013 11:50 AM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] Passing credentials when invoking "curl"*** >> * >> >> ** ** >> >> I am trying to use curl from a lua script to get data from a webservice >> and need to pass user/password information. Can someone help me with this? >> Here is what I am doing, which does not seem to work.**** >> >> ** ** >> >> api = freeswitch.API()**** >> >> agent_status_url = "-u userid:password >> http://mywebservice.com:8080/status/agent/test"**** >> >> response = api:execute("curl", agent_status_url)**** >> >> freeswitch.consoleLog("INFO", response)**** >> >> ** ** >> >> Thanks**** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/889944cb/attachment.html From msc at freeswitch.org Fri Jan 25 23:47:14 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 12:47:14 -0800 Subject: [Freeswitch-users] Freeswitch TLS certificate control In-Reply-To: References: Message-ID: Try setting it to "none" and see if that gets you anywhere. -MC On Thu, Jan 24, 2013 at 8:03 AM, Burak ?zt?rk wrote: > tls-verify-policy -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/e2caaa25/attachment.html From royj at yandex.ru Sat Jan 26 00:26:28 2013 From: royj at yandex.ru (royj) Date: Sat, 26 Jan 2013 01:26:28 +0400 Subject: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error In-Reply-To: References: <965759A53E43FE439E43565A7715E5F0591C085DB2@oa-exchange1.oa.com.au> Message-ID: <20130126012628.ef48f3c88e1b57228db2835c@yandex.ru> I also interested On Fri, 25 Jan 2013 12:09:27 -0800 Michael Collins wrote: > I confess that I haven't seen this specific error before. Could you get the > SIP trace of this call and put it on pastebin as well? That might give you > the information you need to crack the case. > -MC > > On Thu, Jan 24, 2013 at 10:41 PM, Sirish Masur Mohan < > Sirish.MasurMohan at oa.com.au> wrote: > > > Hello All,**** > > > > ** ** > > > > I think I am missing something obvious here, but I am having trouble > > trying to make calls between two SNOM phones. I have an open ACL for > > registration and I keep getting BEARERCAPABILITY_NOTAUTH error when I try > > to make calls. **** > > > > ** ** > > > > Would appreciate any inputs on how to debug this! Logs are at : > > http://pastebin.freeswitch.org/20504**** > > > > ** ** > > > > With regards,**** > > > > Sirish**** > > > > ** ** > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org -- From jaykris at gmail.com Sat Jan 26 00:56:29 2013 From: jaykris at gmail.com (JP) Date: Fri, 25 Jan 2013 13:56:29 -0800 Subject: [Freeswitch-users] Passing credentials when invoking "curl" In-Reply-To: References: <000601cdfa6e$159a6940$40cf3bc0$@center.com> Message-ID: MC, I sure am willing to pay the wiki tax :-). Since I am pretty new to this stuff, can you point me to some place that explains what I need to do? Thanks JP On Fri, Jan 25, 2013 at 12:29 PM, Michael Collins wrote: > Awesome! Now it's time to pay the wiki tax! :) > > -MC > > > On Thu, Jan 24, 2013 at 1:45 PM, JP wrote: > >> Thanks Gilbert. That worked for me. >> -JP >> >> On Thu, Jan 24, 2013 at 12:04 PM, Gilbert Amar wrote: >> >>> Hi **** >>> >>> If it is a basic authentication, you can try **** >>> >>> http://userid:password at mywebservice.com:8080/status/agent/test**** >>> >>> ** ** >>> >>> not sure it works try first a direct from command line :**** >>> >>> ** ** >>> >>> curl "http://userid:password at mywebservice.com:8080/status/agent/test" ** >>> ** >>> >>> ** ** >>> >>> Gilbert**** >>> >>> ** ** >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *JP >>> *Sent:* Thursday, January 24, 2013 11:50 AM >>> *To:* FreeSWITCH Users Help >>> *Subject:* [Freeswitch-users] Passing credentials when invoking "curl"** >>> ** >>> >>> ** ** >>> >>> I am trying to use curl from a lua script to get data from a webservice >>> and need to pass user/password information. Can someone help me with this? >>> Here is what I am doing, which does not seem to work.**** >>> >>> ** ** >>> >>> api = freeswitch.API()**** >>> >>> agent_status_url = "-u userid:password >>> http://mywebservice.com:8080/status/agent/test"**** >>> >>> response = api:execute("curl", agent_status_url)**** >>> >>> freeswitch.consoleLog("INFO", response)**** >>> >>> ** ** >>> >>> Thanks**** >>> >>> ** ** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/c78e7489/attachment-0001.html From schoch+freeswitch.org at xwin32.com Sat Jan 26 01:38:52 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Fri, 25 Jan 2013 14:38:52 -0800 Subject: [Freeswitch-users] Passing credentials when invoking "curl" In-Reply-To: References: <000601cdfa6e$159a6940$40cf3bc0$@center.com> Message-ID: On Fri, Jan 25, 2013 at 1:56 PM, JP wrote: > I sure am willing to pay the wiki tax :-). Since I am pretty new to this > stuff, can you point me to some place that explains what I need to do? > Go to http://www.freeswitch.org/ and click on the "Donate" button. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/b7f77280/attachment.html From jaykris at gmail.com Sat Jan 26 02:00:17 2013 From: jaykris at gmail.com (JP) Date: Fri, 25 Jan 2013 15:00:17 -0800 Subject: [Freeswitch-users] Executing multiple applications in parallel In-Reply-To: References: Message-ID: Avi, Thanks for the response. Unfortunately that did not work. This is what I did The music only plays after the Lua script exits. I played around with the timer value, but that did not help either. Any other ideas? Thanks JP On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: > ESL gives you more control, especially to execute things in parallel. > If you want to do it within FS, that's a little more complicated, since > most things are sequential. > > However, if you use the async APIs, then I think it could work: > 1) sched_broadcast -- > schedule the playback of your file on the UUID of the channel, for +0, > meaning now. > 2) run your lua script > 3) Then stop (break) the playback on that uuid: break > 4) Continue with your execution. > > Please wikify your results and let us know how that works out! > -Avi > > On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: > >> I want to do the following... >> >> 1. Play a long music file from a dialplan. >> 2. While the music is playing, I want to launch a Lua script to do some >> data dip. >> 3. Once the data is fetched and Lua exits, I want to interrupt the music >> and continue in the dialplan >> >> How do I do this. Any sample code or pointers would be greatly >> appreciated. >> >> Thanks >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/2e080b8f/attachment.html From yungwei at resolvity.com Sat Jan 26 02:06:41 2013 From: yungwei at resolvity.com (Yungwei Chen) Date: Fri, 25 Jan 2013 18:06:41 -0500 Subject: [Freeswitch-users] voices in the recordings are out of sync In-Reply-To: References: <33095823FD21DF429B481B5163264B799F6D8752B6@VMBX102.ihostexchange.net> Message-ID: <33095823FD21DF429B481B5163264B799F879B1B1A@VMBX102.ihostexchange.net> Hi, I did a quick test against freeswitch-1.2.5.3 on CentOS 5.8, but the problem is still there. In the recording, the caller speaks way faster than he did. Here's the version of freeswitch: FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z) Here's the set of steps: 1. In dialplan/default/resolvity.xml, add the following extension: 2. Create scripts/test.js with the following content. This js file will read numbers sequentially starting from 0. session.execute("record_session", "/tmp/test.wav"); for (var i=0; i<100; i++) { session.speak("flite", "kal", i+''); } session.hangup(16); 3. reloadxml 4. dial 0009 from a registered SIP client, and then repeat each number you heard. 5. Listen to the recording, /tmp/test.wav. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, December 18, 2012 8:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] voices in the recordings are out of sync Latest version of FreeSWITCH has some updates that may fix this issue. I would update to 1.2.5.3 ASAP. -MC On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen > wrote: Hi, I found one issue that voices are always out of sync in the recordings. I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed from yum. I am having trouble installing the latest version from source due to an error: Autoconf version 2.62 or higher is required. It would be nice if someone can reproduce this issue against HEAD. Thanks. Here're the steps to reproduce it. The idea is to call a phone number and then bridge to another phone number while the entire session is being recorded. 1. In dialplan/public.xml, make sure you have a dialplan to handle any 10 digit phone numbers. 2. In dialplan/default/main.xml, make sure you have an extension to handle the call in the default context. 3. In sip_profiles/external/gateways.xml, make sure you have a gateway that allows you to make an outbound call. 4. make a call to one of the allowed 10-digit phone numbers in your environment. 5. Once the call is answered, the caller shall start to count from 1 to 60 with some pause after each number. 6. The callee shall repeat each number he/she heard from the caller. 7. You should be able to hear that 2 voices in the recoridng (/tmp/rec.wav) are out of sync. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/23b380af/attachment-0001.html From msc at freeswitch.org Sat Jan 26 02:27:47 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 15:27:47 -0800 Subject: [Freeswitch-users] Passing credentials when invoking "curl" In-Reply-To: References: <000601cdfa6e$159a6940$40cf3bc0$@center.com> Message-ID: Also, wiki.freeswitch.org Sign up for an account. Then go to the page where you were looking for this information and edit the page to include this information. Let me know if you have any questions. -MC On Fri, Jan 25, 2013 at 2:38 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > On Fri, Jan 25, 2013 at 1:56 PM, JP wrote: > >> I sure am willing to pay the wiki tax :-). Since I am pretty new to this >> stuff, can you point me to some place that explains what I need to do? >> > > Go to http://www.freeswitch.org/ and click on the "Donate" button. > > -- > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/738382a8/attachment.html From jaykris at gmail.com Sat Jan 26 02:40:29 2013 From: jaykris at gmail.com (JP) Date: Fri, 25 Jan 2013 15:40:29 -0800 Subject: [Freeswitch-users] Passing credentials when invoking "curl" In-Reply-To: References: <000601cdfa6e$159a6940$40cf3bc0$@center.com> Message-ID: Thanks, I have requested for an account. As soon as I get it I will update the mod-curl page with this info. -JP On Fri, Jan 25, 2013 at 3:27 PM, Michael Collins wrote: > Also, > > wiki.freeswitch.org > > Sign up for an account. Then go to the page where you were looking for > this information and edit the page to include this information. > > Let me know if you have any questions. > > -MC > > On Fri, Jan 25, 2013 at 2:38 PM, Steven Schoch < > schoch+freeswitch.org at xwin32.com> wrote: > >> On Fri, Jan 25, 2013 at 1:56 PM, JP wrote: >> >>> I sure am willing to pay the wiki tax :-). Since I am pretty new to this >>> stuff, can you point me to some place that explains what I need to do? >>> >> >> Go to http://www.freeswitch.org/ and click on the "Donate" button. >> >> -- >> Steve >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/3e9f6509/attachment.html From msc at freeswitch.org Sat Jan 26 03:01:16 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 16:01:16 -0800 Subject: [Freeswitch-users] Passing credentials when invoking "curl" In-Reply-To: References: <000601cdfa6e$159a6940$40cf3bc0$@center.com> Message-ID: Okay, go for it! Thanks, MC On Fri, Jan 25, 2013 at 3:40 PM, JP wrote: > Thanks, I have requested for an account. As soon as I get it I will update > the mod-curl page with this info. > > -JP > > On Fri, Jan 25, 2013 at 3:27 PM, Michael Collins wrote: > >> Also, >> >> wiki.freeswitch.org >> >> Sign up for an account. Then go to the page where you were looking for >> this information and edit the page to include this information. >> >> Let me know if you have any questions. >> >> -MC >> >> On Fri, Jan 25, 2013 at 2:38 PM, Steven Schoch < >> schoch+freeswitch.org at xwin32.com> wrote: >> >>> On Fri, Jan 25, 2013 at 1:56 PM, JP wrote: >>> >>>> I sure am willing to pay the wiki tax :-). Since I am pretty new to >>>> this stuff, can you point me to some place that explains what I need to do? >>>> >>> >>> Go to http://www.freeswitch.org/ and click on the "Donate" button. >>> >>> -- >>> Steve >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/810f45f5/attachment-0001.html From msc at freeswitch.org Sat Jan 26 03:10:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 16:10:37 -0800 Subject: [Freeswitch-users] Executing multiple applications in parallel In-Reply-To: References: Message-ID: The problem you're running into is that you are trying to do two different things inside of the thread that is controlling the call flow. Playing sound files and running Lua scripts are synchronous in this context. This seems more suited for an outbound event socket application. Using the event socket you can control the call while doing other things. Just curious - does anyone have an ESL example of this kind of thing? The wiki could definitely use this kind of example as it is quite useful. I'll look around and see if I can find anything. -MC On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: > Avi, > Thanks for the response. Unfortunately that did not work. This is what I > did > > > > > > > > > > > > > The music only plays after the Lua script exits. I played around with the > timer value, but that did not help either. Any other ideas? > > Thanks > JP > > > On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: > >> ESL gives you more control, especially to execute things in parallel. >> If you want to do it within FS, that's a little more complicated, since >> most things are sequential. >> >> However, if you use the async APIs, then I think it could work: >> 1) sched_broadcast -- >> schedule the playback of your file on the UUID of the channel, for +0, >> meaning now. >> 2) run your lua script >> 3) Then stop (break) the playback on that uuid: break >> 4) Continue with your execution. >> >> Please wikify your results and let us know how that works out! >> -Avi >> >> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >> >>> I want to do the following... >>> >>> 1. Play a long music file from a dialplan. >>> 2. While the music is playing, I want to launch a Lua script to do some >>> data dip. >>> 3. Once the data is fetched and Lua exits, I want to interrupt the music >>> and continue in the dialplan >>> >>> How do I do this. Any sample code or pointers would be greatly >>> appreciated. >>> >>> Thanks >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/8bcf7b17/attachment.html From jaykris at gmail.com Sat Jan 26 03:32:50 2013 From: jaykris at gmail.com (JP) Date: Fri, 25 Jan 2013 16:32:50 -0800 Subject: [Freeswitch-users] Executing multiple applications in parallel In-Reply-To: References: Message-ID: MC, Thanks for the info. Unfortunately outbound event socket application is not an option for me right now. I was wondering if the following "ideas" can be of any use to me( I know I am trying clutch at straws, but hope is eternal :)) 1. Use "bgapi". Although on the wiki I only see this used with "originate". 2. Can I use a conference to park the caller there while playing music on another call leg of the conference. Meanwhile can I execute the Lua script :)) 3. Can I launch another Lua script from my original one asynchronously? 4. Can I have a start up Lua script running in the background listening for some custom events and I can have the main caller thread generate the custom event and then start playing long music file. Once the background Lua script is done with the data dip, it can just break the music on the other channel. That's it from me. Someone help me from sinking... please. Thanks, -JP On Fri, Jan 25, 2013 at 4:10 PM, Michael Collins wrote: > The problem you're running into is that you are trying to do two different > things inside of the thread that is controlling the call flow. Playing > sound files and running Lua scripts are synchronous in this context. This > seems more suited for an outbound event socket application. Using the event > socket you can control the call while doing other things. > > Just curious - does anyone have an ESL example of this kind of thing? The > wiki could definitely use this kind of example as it is quite useful. I'll > look around and see if I can find anything. > > -MC > > > On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: > >> Avi, >> Thanks for the response. Unfortunately that did not work. This is what I >> did >> >> >> >> >> >> >> >> >> >> >> >> >> The music only plays after the Lua script exits. I played around with the >> timer value, but that did not help either. Any other ideas? >> >> Thanks >> JP >> >> >> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >> >>> ESL gives you more control, especially to execute things in parallel. >>> If you want to do it within FS, that's a little more complicated, since >>> most things are sequential. >>> >>> However, if you use the async APIs, then I think it could work: >>> 1) sched_broadcast -- >>> schedule the playback of your file on the UUID of the channel, for +0, >>> meaning now. >>> 2) run your lua script >>> 3) Then stop (break) the playback on that uuid: break >>> 4) Continue with your execution. >>> >>> Please wikify your results and let us know how that works out! >>> -Avi >>> >>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>> >>>> I want to do the following... >>>> >>>> 1. Play a long music file from a dialplan. >>>> 2. While the music is playing, I want to launch a Lua script to do some >>>> data dip. >>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>> music and continue in the dialplan >>>> >>>> How do I do this. Any sample code or pointers would be greatly >>>> appreciated. >>>> >>>> Thanks >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/dbe8716f/attachment-0001.html From itsusama at gmail.com Sat Jan 26 03:44:10 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Sat, 26 Jan 2013 05:44:10 +0500 Subject: [Freeswitch-users] Executing Multiple Applications in parallel Message-ID: Hey, I posted this earlier as a question, but I think this might be relevant. ttp://pastebin.freeswitch.org/20508 . The workaround (hack) for my issue is to check the connection state ($con->connected()) every time you loop and reconnect and subscribe for events again when that returns false. On Sat, Jan 26, 2013 at 5:33 AM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Executing multiple applications in parallel (Michael Collins) > 2. Re: Executing multiple applications in parallel (JP) > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Fri, 25 Jan 2013 16:10:37 -0800 > Subject: Re: [Freeswitch-users] Executing multiple applications in parallel > The problem you're running into is that you are trying to do two different > things inside of the thread that is controlling the call flow. Playing > sound files and running Lua scripts are synchronous in this context. This > seems more suited for an outbound event socket application. Using the event > socket you can control the call while doing other things. > > Just curious - does anyone have an ESL example of this kind of thing? The > wiki could definitely use this kind of example as it is quite useful. I'll > look around and see if I can find anything. > > -MC > > On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: > >> Avi, >> Thanks for the response. Unfortunately that did not work. This is what I >> did >> >> >> >> >> >> >> >> >> >> >> >> >> The music only plays after the Lua script exits. I played around with the >> timer value, but that did not help either. Any other ideas? >> >> Thanks >> JP >> >> >> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >> >>> ESL gives you more control, especially to execute things in parallel. >>> If you want to do it within FS, that's a little more complicated, since >>> most things are sequential. >>> >>> However, if you use the async APIs, then I think it could work: >>> 1) sched_broadcast -- >>> schedule the playback of your file on the UUID of the channel, for +0, >>> meaning now. >>> 2) run your lua script >>> 3) Then stop (break) the playback on that uuid: break >>> 4) Continue with your execution. >>> >>> Please wikify your results and let us know how that works out! >>> -Avi >>> >>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>> >>>> I want to do the following... >>>> >>>> 1. Play a long music file from a dialplan. >>>> 2. While the music is playing, I want to launch a Lua script to do some >>>> data dip. >>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>> music and continue in the dialplan >>>> >>>> How do I do this. Any sample code or pointers would be greatly >>>> appreciated. >>>> >>>> Thanks >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: JP > To: FreeSWITCH Users Help > Cc: > Date: Fri, 25 Jan 2013 16:32:50 -0800 > Subject: Re: [Freeswitch-users] Executing multiple applications in parallel > MC, > Thanks for the info. Unfortunately outbound event socket application is > not an option for me right now. I was wondering if the following "ideas" > can be of any use to me( I know I am trying clutch at straws, but hope is > eternal :)) > 1. Use "bgapi". Although on the wiki I only see this used with "originate". > 2. Can I use a conference to park the caller there while playing music on > another call leg of the conference. Meanwhile can I execute the Lua script > :)) > 3. Can I launch another Lua script from my original one asynchronously? > 4. Can I have a start up Lua script running in the background listening > for some custom events and I can have the main caller thread generate the > custom event and then start playing long music file. Once the background > Lua script is done with the data dip, it can just break the music on the > other channel. > > That's it from me. Someone help me from sinking... please. > > Thanks, > -JP > > > On Fri, Jan 25, 2013 at 4:10 PM, Michael Collins wrote: > >> The problem you're running into is that you are trying to do two >> different things inside of the thread that is controlling the call flow. >> Playing sound files and running Lua scripts are synchronous in this >> context. This seems more suited for an outbound event socket application. >> Using the event socket you can control the call while doing other things. >> >> Just curious - does anyone have an ESL example of this kind of thing? The >> wiki could definitely use this kind of example as it is quite useful. I'll >> look around and see if I can find anything. >> >> -MC >> >> >> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >> >>> Avi, >>> Thanks for the response. Unfortunately that did not work. This is what I >>> did >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> The music only plays after the Lua script exits. I played around with >>> the timer value, but that did not help either. Any other ideas? >>> >>> Thanks >>> JP >>> >>> >>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>> >>>> ESL gives you more control, especially to execute things in parallel. >>>> If you want to do it within FS, that's a little more complicated, since >>>> most things are sequential. >>>> >>>> However, if you use the async APIs, then I think it could work: >>>> 1) sched_broadcast -- >>>> schedule the playback of your file on the UUID of the channel, for +0, >>>> meaning now. >>>> 2) run your lua script >>>> 3) Then stop (break) the playback on that uuid: break >>>> 4) Continue with your execution. >>>> >>>> Please wikify your results and let us know how that works out! >>>> -Avi >>>> >>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>> >>>>> I want to do the following... >>>>> >>>>> 1. Play a long music file from a dialplan. >>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>> some data dip. >>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>> music and continue in the dialplan >>>>> >>>>> How do I do this. Any sample code or pointers would be greatly >>>>> appreciated. >>>>> >>>>> Thanks >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130126/4018ed7d/attachment-0001.html From jaykris at gmail.com Sat Jan 26 04:02:20 2013 From: jaykris at gmail.com (JP) Date: Fri, 25 Jan 2013 17:02:20 -0800 Subject: [Freeswitch-users] Executing Multiple Applications in parallel In-Reply-To: References: Message-ID: Hi, I am not able to access the link you pasted. Can you please paste the workaround in this thread? Thanks JP On Fri, Jan 25, 2013 at 4:44 PM, Usama Zaidi wrote: > Hey, > > I posted this earlier as a question, but I think this might be relevant. > ttp://pastebin.freeswitch.org/20508 . > The workaround (hack) for my issue is to check the connection state > ($con->connected()) every time you loop and reconnect and subscribe for > events again when that returns false. > > On Sat, Jan 26, 2013 at 5:33 AM, < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: Executing multiple applications in parallel (Michael Collins) >> 2. Re: Executing multiple applications in parallel (JP) >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins >> To: FreeSWITCH Users Help >> Cc: >> Date: Fri, 25 Jan 2013 16:10:37 -0800 >> Subject: Re: [Freeswitch-users] Executing multiple applications in >> parallel >> The problem you're running into is that you are trying to do two >> different things inside of the thread that is controlling the call flow. >> Playing sound files and running Lua scripts are synchronous in this >> context. This seems more suited for an outbound event socket application. >> Using the event socket you can control the call while doing other things. >> >> Just curious - does anyone have an ESL example of this kind of thing? The >> wiki could definitely use this kind of example as it is quite useful. I'll >> look around and see if I can find anything. >> >> -MC >> >> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >> >>> Avi, >>> Thanks for the response. Unfortunately that did not work. This is what I >>> did >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> The music only plays after the Lua script exits. I played around with >>> the timer value, but that did not help either. Any other ideas? >>> >>> Thanks >>> JP >>> >>> >>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>> >>>> ESL gives you more control, especially to execute things in parallel. >>>> If you want to do it within FS, that's a little more complicated, since >>>> most things are sequential. >>>> >>>> However, if you use the async APIs, then I think it could work: >>>> 1) sched_broadcast -- >>>> schedule the playback of your file on the UUID of the channel, for +0, >>>> meaning now. >>>> 2) run your lua script >>>> 3) Then stop (break) the playback on that uuid: break >>>> 4) Continue with your execution. >>>> >>>> Please wikify your results and let us know how that works out! >>>> -Avi >>>> >>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>> >>>>> I want to do the following... >>>>> >>>>> 1. Play a long music file from a dialplan. >>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>> some data dip. >>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>> music and continue in the dialplan >>>>> >>>>> How do I do this. Any sample code or pointers would be greatly >>>>> appreciated. >>>>> >>>>> Thanks >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> ---------- Forwarded message ---------- >> From: JP >> To: FreeSWITCH Users Help >> Cc: >> Date: Fri, 25 Jan 2013 16:32:50 -0800 >> Subject: Re: [Freeswitch-users] Executing multiple applications in >> parallel >> MC, >> Thanks for the info. Unfortunately outbound event socket application is >> not an option for me right now. I was wondering if the following "ideas" >> can be of any use to me( I know I am trying clutch at straws, but hope is >> eternal :)) >> 1. Use "bgapi". Although on the wiki I only see this used with >> "originate". >> 2. Can I use a conference to park the caller there while playing music on >> another call leg of the conference. Meanwhile can I execute the Lua script >> :)) >> 3. Can I launch another Lua script from my original one asynchronously? >> 4. Can I have a start up Lua script running in the background listening >> for some custom events and I can have the main caller thread generate the >> custom event and then start playing long music file. Once the background >> Lua script is done with the data dip, it can just break the music on the >> other channel. >> >> That's it from me. Someone help me from sinking... please. >> >> Thanks, >> -JP >> >> >> On Fri, Jan 25, 2013 at 4:10 PM, Michael Collins wrote: >> >>> The problem you're running into is that you are trying to do two >>> different things inside of the thread that is controlling the call flow. >>> Playing sound files and running Lua scripts are synchronous in this >>> context. This seems more suited for an outbound event socket application. >>> Using the event socket you can control the call while doing other things. >>> >>> Just curious - does anyone have an ESL example of this kind of thing? >>> The wiki could definitely use this kind of example as it is quite useful. >>> I'll look around and see if I can find anything. >>> >>> -MC >>> >>> >>> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >>> >>>> Avi, >>>> Thanks for the response. Unfortunately that did not work. This is what >>>> I did >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> The music only plays after the Lua script exits. I played around with >>>> the timer value, but that did not help either. Any other ideas? >>>> >>>> Thanks >>>> JP >>>> >>>> >>>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>>> >>>>> ESL gives you more control, especially to execute things in parallel. >>>>> If you want to do it within FS, that's a little more complicated, >>>>> since most things are sequential. >>>>> >>>>> However, if you use the async APIs, then I think it could work: >>>>> 1) sched_broadcast -- >>>>> schedule the playback of your file on the UUID of the channel, for +0, >>>>> meaning now. >>>>> 2) run your lua script >>>>> 3) Then stop (break) the playback on that uuid: break >>>>> 4) Continue with your execution. >>>>> >>>>> Please wikify your results and let us know how that works out! >>>>> -Avi >>>>> >>>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>>> >>>>>> I want to do the following... >>>>>> >>>>>> 1. Play a long music file from a dialplan. >>>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>>> some data dip. >>>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>>> music and continue in the dialplan >>>>>> >>>>>> How do I do this. Any sample code or pointers would be greatly >>>>>> appreciated. >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > I'd love to change the world, but they wont gimme the source code to it > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/ad7e7a6f/attachment-0001.html From msc at freeswitch.org Sat Jan 26 04:42:23 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 25 Jan 2013 17:42:23 -0800 Subject: [Freeswitch-users] Executing multiple applications in parallel In-Reply-To: References: Message-ID: Well, I did think of one thing that you could try. If it works it would be kind of cute: Answer the call like you currently do, grab the digits from the caller Set the transfer_ringback variable to ${hold_music} (or whatever you want them to hear) Transfer the call to another extension, like this: Create a new extension like this: ... do whatever you need to do with the call after the data dip The theory is that the "please hold while we look up your info..." sound is the "transfer ringback" sound. The target extension "data_dip" then "rings" while it's looking up the info and then "answers" after the lookup is complete. I didn't try it so please remember that the standard disclaimerapplies. Let us know if you make any headway on this. -MC On Fri, Jan 25, 2013 at 4:32 PM, JP wrote: > MC, > Thanks for the info. Unfortunately outbound event socket application is > not an option for me right now. I was wondering if the following "ideas" > can be of any use to me( I know I am trying clutch at straws, but hope is > eternal :)) > 1. Use "bgapi". Although on the wiki I only see this used with "originate". > 2. Can I use a conference to park the caller there while playing music on > another call leg of the conference. Meanwhile can I execute the Lua script > :)) > 3. Can I launch another Lua script from my original one asynchronously? > 4. Can I have a start up Lua script running in the background listening > for some custom events and I can have the main caller thread generate the > custom event and then start playing long music file. Once the background > Lua script is done with the data dip, it can just break the music on the > other channel. > > That's it from me. Someone help me from sinking... please. > > Thanks, > -JP > > > On Fri, Jan 25, 2013 at 4:10 PM, Michael Collins wrote: > >> The problem you're running into is that you are trying to do two >> different things inside of the thread that is controlling the call flow. >> Playing sound files and running Lua scripts are synchronous in this >> context. This seems more suited for an outbound event socket application. >> Using the event socket you can control the call while doing other things. >> >> Just curious - does anyone have an ESL example of this kind of thing? The >> wiki could definitely use this kind of example as it is quite useful. I'll >> look around and see if I can find anything. >> >> -MC >> >> >> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >> >>> Avi, >>> Thanks for the response. Unfortunately that did not work. This is what I >>> did >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> The music only plays after the Lua script exits. I played around with >>> the timer value, but that did not help either. Any other ideas? >>> >>> Thanks >>> JP >>> >>> >>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>> >>>> ESL gives you more control, especially to execute things in parallel. >>>> If you want to do it within FS, that's a little more complicated, since >>>> most things are sequential. >>>> >>>> However, if you use the async APIs, then I think it could work: >>>> 1) sched_broadcast -- >>>> schedule the playback of your file on the UUID of the channel, for +0, >>>> meaning now. >>>> 2) run your lua script >>>> 3) Then stop (break) the playback on that uuid: break >>>> 4) Continue with your execution. >>>> >>>> Please wikify your results and let us know how that works out! >>>> -Avi >>>> >>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>> >>>>> I want to do the following... >>>>> >>>>> 1. Play a long music file from a dialplan. >>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>> some data dip. >>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>> music and continue in the dialplan >>>>> >>>>> How do I do this. Any sample code or pointers would be greatly >>>>> appreciated. >>>>> >>>>> Thanks >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/44178320/attachment.html From jaykris at gmail.com Sat Jan 26 05:29:05 2013 From: jaykris at gmail.com (JP) Date: Fri, 25 Jan 2013 18:29:05 -0800 Subject: [Freeswitch-users] Executing multiple applications in parallel In-Reply-To: References: Message-ID: Just my luck. It didn't work :( Here is what I did... The following is the logs from FS 2013-01-25 18:20:59.210491 [DEBUG] switch_rtp.c:1927 Starting timer [soft] 160 bytes per 20ms 2013-01-25 18:20:59.210491 [DEBUG] sofia_glue.c:3609 Set 2833 dtmf send payload to 101 2013-01-25 18:20:59.210491 [DEBUG] sofia_glue.c:3615 Set 2833 dtmf receive payload to 101 2013-01-25 18:20:59.210491 [DEBUG] sofia_glue.c:3642 sofia/external/ 16502928610 at 10.191.210.148 Set rtp dtmf delay to 40 2013-01-25 18:20:59.210491 [NOTICE] sofia_glue.c:4253 Pre-Answer sofia/external/16502928610 at 10.191.210.148! 2013-01-25 18:20:59.210491 [DEBUG] switch_channel.c:3083 (sofia/external/ 16502928610 at 10.191.210.148) Callstate Change RINGING -> EARLY 2013-01-25 18:20:59.210491 [DEBUG] mod_sofia.c:854 Local SDP sofia/external/ 16502928610 at 10.191.210.148: v=0 o=FreeSWITCH 1359139145 1359139146 IN IP4 10.191.210.141 s=FreeSWITCH c=IN IP4 10.191.210.141 t=0 0 m=audio 27714 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2013-01-25 18:20:59.210491 [DEBUG] switch_core_session.c:823 Send signal sofia/external/16502928610 at 10.191.210.148 [BREAK] 2013-01-25 18:20:59.210491 [DEBUG] switch_channel.c:3342 (sofia/external/ 16502928610 at 10.191.210.148) Callstate Change EARLY -> ACTIVE 2013-01-25 18:20:59.210491 [NOTICE] mod_dptools.c:1176 Channel [sofia/external/16502928610 at 10.191.210.148] has been answered 2013-01-25 18:20:59.210491 [DEBUG] switch_core_session.c:969 Send signal sofia/external/16502928610 at 10.191.210.148 [BREAK] 2013-01-25 18:20:59.210491 [DEBUG] sofia.c:5591 Channel sofia/external/ 16502928610 at 10.191.210.148 entering state [completed][200] EXECUTE sofia/external/16502928610 at 10.191.210.148set(transfer_ringback=/tmp/WPJ_MUSIC_QUEUE_4.wav) 2013-01-25 18:20:59.210491 [DEBUG] mod_dptools.c:1344 sofia/external/ 16502928610 at 10.191.210.148 SET [transfer_ringback]=[/tmp/WPJ_MUSIC_QUEUE_4.wav] EXECUTE sofia/external/16502928610 at 10.191.210.148 set(tts_engine=flite) 2013-01-25 18:20:59.210491 [DEBUG] mod_dptools.c:1344 sofia/external/ 16502928610 at 10.191.210.148 SET [tts_engine]=[flite] EXECUTE sofia/external/16502928610 at 10.191.210.148 set(tts_voice=kal) 2013-01-25 18:20:59.210491 [DEBUG] mod_dptools.c:1344 sofia/external/ 16502928610 at 10.191.210.148 SET [tts_voice]=[kal] EXECUTE sofia/external/16502928610 at 10.191.210.148 play_and_get_digits(4 16 3 7000 # phrase:Callidus_OD_Prompt phrase:Callidus_invalid_entry case_number \d+) 2013-01-25 18:20:59.210491 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2013-01-25 18:20:59.231540 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_the.wav] (en:en) 2013-01-25 18:20:59.231540 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms 2013-01-25 18:20:59.231540 [DEBUG] switch_core_session.c:969 Send signal sofia/external/16502928610 at 10.191.210.148 [BREAK] 2013-01-25 18:20:59.231540 [DEBUG] switch_core_session.c:969 Send signal sofia/external/16502928610 at 10.191.210.148 [BREAK] 2013-01-25 18:20:59.231540 [DEBUG] switch_core_session.c:969 Send signal sofia/external/16502928610 at 10.191.210.148 [BREAK] 2013-01-25 18:20:59.250478 [DEBUG] sofia.c:5591 Channel sofia/external/ 16502928610 at 10.191.210.148 entering state [ready][200] 2013-01-25 18:20:59.291532 [DEBUG] switch_rtp.c:3606 Correct ip/port confirmed. 2013-01-25 18:20:59.950314 [DEBUG] switch_ivr_play_say.c:1682 done playing file /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_the.wav 2013-01-25 18:21:00.010301 [DEBUG] switch_ivr_play_say.c:244 Handle speak-text:[case number,] (en:en) 2013-01-25 18:21:00.010301 [DEBUG] switch_ivr_play_say.c:2481 OPEN TTS flite 2013-01-25 18:21:00.010301 [DEBUG] switch_ivr_play_say.c:2490 Raw Codec Activated 2013-01-25 18:21:00.030327 [DEBUG] switch_ivr_play_say.c:2172 Speaking text: case number, 2013-01-25 18:21:01.710910 [DEBUG] switch_ivr_play_say.c:2369 done speaking text 2013-01-25 18:21:01.770904 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-followed_by_pound.wav] (en:en) 2013-01-25 18:21:01.770904 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms 2013-01-25 18:21:03.690471 [DEBUG] switch_ivr_play_say.c:1682 done playing file /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-followed_by_pound.wav 2013-01-25 18:21:04.350311 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 3:800 2013-01-25 18:21:04.790211 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 0:800 2013-01-25 18:21:05.250108 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 0:800 2013-01-25 18:21:05.690999 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 0:800 2013-01-25 18:21:06.049934 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 3:800 2013-01-25 18:21:06.470825 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 1:800 2013-01-25 18:21:06.790751 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 8:800 2013-01-25 18:21:07.191718 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF 4:800 2013-01-25 18:21:08.230423 [DEBUG] switch_rtp.c:3809 RTP RECV DTMF #:800 2013-01-25 18:21:08.230423 [DEBUG] switch_ivr_play_say.c:2034 Test Regex [30003184][\d+] EXECUTE sofia/external/16502928610 at 10.191.210.148 transfer(data_dip) 2013-01-25 18:21:08.230423 [DEBUG] switch_ivr.c:1761 (sofia/external/ 16502928610 at 10.191.210.148) State Change CS_EXECUTE -> CS_ROUTING 2013-01-25 18:21:08.230423 [DEBUG] switch_core_session.c:1275 Send signal sofia/external/16502928610 at 10.191.210.148 [BREAK] 2013-01-25 18:21:08.230423 [DEBUG] switch_core_session.c:823 Send signal sofia/external/16502928610 at 10.191.210.148 [BREAK] 2013-01-25 18:21:08.230423 [NOTICE] switch_ivr.c:1767 Transfer sofia/external/16502928610 at 10.191.210.148 to XML[data_dip at public] 2013-01-25 18:21:08.230423 [DEBUG] switch_core_state_machine.c:477 (sofia/external/16502928610 at 10.191.210.148) State EXECUTE going to sleep 2013-01-25 18:21:08.230423 [DEBUG] switch_core_state_machine.c:415 (sofia/external/16502928610 at 10.191.210.148) Running State Change CS_ROUTING 2013-01-25 18:21:08.230423 [DEBUG] switch_channel.c:1960 (sofia/external/ 16502928610 at 10.191.210.148) Callstate Change ACTIVE -> RINGING 2013-01-25 18:21:08.230423 [DEBUG] switch_core_state_machine.c:470 (sofia/external/16502928610 at 10.191.210.148) State ROUTING 2013-01-25 18:21:08.230423 [DEBUG] mod_sofia.c:149 sofia/external/ 16502928610 at 10.191.210.148 SOFIA ROUTING 2013-01-25 18:21:08.230423 [DEBUG] switch_core_state_machine.c:117 sofia/external/16502928610 at 10.191.210.148 Standard ROUTING 2013-01-25 18:21:08.230423 [INFO] mod_dialplan_xml.c:497 Processing dev7jptest#16502928610#X <16502928610>->data_dip in context public Dialplan: sofia/external/16502928610 at 10.191.210.148 parsing [public->unloop] continue=false Dialplan: sofia/external/16502928610 at 10.191.210.148 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/16502928610 at 10.191.210.148 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/16502928610 at 10.191.210.148 parsing [public->outside_call] continue=true Dialplan: sofia/external/16502928610 at 10.191.210.148 Absolute Condition [outside_call] Dialplan: sofia/external/16502928610 at 10.191.210.148 Action set(outside_call=true) Dialplan: sofia/external/16502928610 at 10.191.210.148 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/16502928610 at 10.191.210.148 parsing [public->call_debug] continue=true Dialplan: sofia/external/16502928610 at 10.191.210.148 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/16502928610 at 10.191.210.148 parsing [public->public_extensions] continue=false Dialplan: sofia/external/16502928610 at 10.191.210.148 Regex (FAIL) [public_extensions] destination_number(data_dip) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/16502928610 at 10.191.210.148 parsing [public->public_did] continue=false Dialplan: sofia/external/16502928610 at 10.191.210.148 Regex (FAIL) [public_did] destination_number(data_dip) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/16502928610 at 10.191.210.148 parsing [public->9010] continue=false Dialplan: sofia/external/16502928610 at 10.191.210.148 Regex (FAIL) [9010] destination_number(data_dip) =~ /^(9010)$/ break=on-false Dialplan: sofia/external/16502928610 at 10.191.210.148 parsing [public->Callidus_OnDemand_IVR] continue=false Dialplan: sofia/external/16502928610 at 10.191.210.148 Regex (FAIL) [Callidus_OnDemand_IVR] destination_number(data_dip) =~ /^(9005)$/ break=on-false Dialplan: sofia/external/16502928610 at 10.191.210.148 parsing [public->Callidus_OnPremise_IVR] continue=false Dialplan: sofia/external/16502928610 at 10.191.210.148 Regex (FAIL) [Callidus_OnPremise_IVR] destination_number(data_dip) =~ /^(9006)$/ break=on-false Dialplan: sofia/external/16502928610 at 10.191.210.148 parsing [public->MY TEST] continue=false Dialplan: sofia/external/16502928610 at 10.191.210.148 Regex (FAIL) [MY TEST] destination_number(data_dip) =~ /8027/ break=on-false Dialplan: sofia/external/16502928610 at 10.191.210.148 parsing [public->data dip] continue=false Dialplan: sofia/external/16502928610 at 10.191.210.148 Regex (PASS) [data dip] destination_number(data_dip) =~ /^data_dip$/ break=on-false Dialplan: sofia/external/16502928610 at 10.191.210.148 Action pre_answer() Dialplan: sofia/external/16502928610 at 10.191.210.148 Action lua(callidus_get_customer_info.lua ${case_number} OnDemand) Dialplan: sofia/external/16502928610 at 10.191.210.148 Action answer() 2013-01-25 18:21:08.230423 [DEBUG] switch_core_state_machine.c:167 (sofia/external/16502928610 at 10.191.210.148) State Change CS_ROUTING -> CS_EXECUTE 2013-01-25 18:21:08.230423 [DEBUG] switch_core_session.c:1275 Send signal sofia/external/16502928610 at 10.191.210.148 [BREAK] 2013-01-25 18:21:08.230423 [DEBUG] switch_core_state_machine.c:470 (sofia/external/16502928610 at 10.191.210.148) State ROUTING going to sleep 2013-01-25 18:21:08.230423 [DEBUG] switch_core_state_machine.c:415 (sofia/external/16502928610 at 10.191.210.148) Running State Change CS_EXECUTE 2013-01-25 18:21:08.230423 [DEBUG] switch_channel.c:1962 (sofia/external/ 16502928610 at 10.191.210.148) Callstate Change RINGING -> ACTIVE 2013-01-25 18:21:08.230423 [DEBUG] switch_core_state_machine.c:477 (sofia/external/16502928610 at 10.191.210.148) State EXECUTE 2013-01-25 18:21:08.230423 [DEBUG] mod_sofia.c:242 sofia/external/ 16502928610 at 10.191.210.148 SOFIA EXECUTE 2013-01-25 18:21:08.230423 [DEBUG] switch_core_state_machine.c:209 sofia/external/16502928610 at 10.191.210.148 Standard EXECUTE EXECUTE sofia/external/16502928610 at 10.191.210.148 set(outside_call=true) 2013-01-25 18:21:08.230423 [DEBUG] mod_dptools.c:1344 sofia/external/ 16502928610 at 10.191.210.148 SET [outside_call]=[true] EXECUTE sofia/external/16502928610 at 10.191.210.148 export(RFC2822_DATE=Fri, 25 Jan 2013 18:21:08 -0800) 2013-01-25 18:21:08.230423 [DEBUG] switch_channel.c:1118 EXPORT (export_vars) [RFC2822_DATE]=[Fri, 25 Jan 2013 18:21:08 -0800] EXECUTE sofia/external/16502928610 at 10.191.210.148 pre_answer() EXECUTE sofia/external/16502928610 at 10.191.210.148lua(callidus_get_customer_info.lua 30003184 OnDemand) 2013-01-25 18:21:08.230423 [DEBUG] mod_curl.c:121 method: get, url: http://vm6devc11.dev.whitepj.net:8080/edsa/object/SupportCase/CaseNumber___SearchStringField%20is%20SC30003184/assigned,Company?agent=dev7jptest~~jp&_type=json 2013-01-25 18:21:14.449993 [DEBUG] mod_curl.c:121 method: get, url: http://vm6devc11.dev.whitepj.net:8080/edsa/object/employee/InternalIdNumber___SearchLongField%20equalTo%20413056/custentity_vcc_agent_id,custentity_vcc_phone_queue_id?agent=dev7jptest~~jp&_type=json 2013-01-25 18:21:20.090845 [DEBUG] mod_curl.c:121 method: get, url: http://dev7jptest:2e3de0389b11796554e2c9dac9717688 at vm6devc11.dev.whitepj.net:8080/tstats/rtstats/agent/test.json 2013-01-25 18:21:22.150301 [DEBUG] switch_cpp.cpp:1007 sofia/external/ 16502928610 at 10.191.210.148 destroy/unlink session from object EXECUTE sofia/external/16502928610 at 10.191.210.148 answer() 2013-01-25 18:21:22.150301 [NOTICE] switch_core_state_machine.c:262 sofia/external/16502928610 at 10.191.210.148 has executed the last dialplan instruction, hanging up. 2013-01-25 18:21:22.150301 [DEBUG] switch_channel.c:2941 (sofia/external/ 16502928610 at 10.191.210.148) Callstate Change ACTIVE -> HANGUP 2013-01-25 18:21:22.150301 [NOTICE] switch_core_state_machine.c:264 Hangup sofia/external/16502928610 at 10.191.210.148 [CS_EXECUTE] [NORMAL_CLEARING] 2013-01-25 18:21:22.150301 [DEBUG] switch_channel.c:2964 Send signal sofia/external/16502928610 at 10.191.210.148 [KILL] 2013-01-25 18:21:22.150301 [DEBUG] switch_core_session.c:1275 Send signal sofia/external/16502928610 at 10.191.210.148 [BREAK] 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:477 (sofia/external/16502928610 at 10.191.210.148) State EXECUTE going to sleep 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:415 (sofia/external/16502928610 at 10.191.210.148) Running State Change CS_HANGUP 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:665 (sofia/external/16502928610 at 10.191.210.148) State HANGUP 2013-01-25 18:21:22.150301 [DEBUG] mod_sofia.c:501 Channel sofia/external/ 16502928610 at 10.191.210.148 hanging up, cause: NORMAL_CLEARING 2013-01-25 18:21:22.150301 [DEBUG] mod_sofia.c:550 Sending BYE to sofia/external/16502928610 at 10.191.210.148 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:48 sofia/external/16502928610 at 10.191.210.148 Standard HANGUP, cause: NORMAL_CLEARING 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:665 (sofia/external/16502928610 at 10.191.210.148) State HANGUP going to sleep 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:446 (sofia/external/16502928610 at 10.191.210.148) State Change CS_HANGUP -> CS_REPORTING 2013-01-25 18:21:22.150301 [DEBUG] switch_core_session.c:1275 Send signal sofia/external/16502928610 at 10.191.210.148 [BREAK] 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:415 (sofia/external/16502928610 at 10.191.210.148) Running State Change CS_REPORTING 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:747 (sofia/external/16502928610 at 10.191.210.148) State REPORTING 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:92 sofia/external/16502928610 at 10.191.210.148 Standard REPORTING, cause: NORMAL_CLEARING 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:747 (sofia/external/16502928610 at 10.191.210.148) State REPORTING going to sleep 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:440 (sofia/external/16502928610 at 10.191.210.148) State Change CS_REPORTING -> CS_DESTROY 2013-01-25 18:21:22.150301 [DEBUG] switch_core_session.c:1275 Send signal sofia/external/16502928610 at 10.191.210.148 [BREAK] 2013-01-25 18:21:22.150301 [DEBUG] switch_core_session.c:1480 Session 64 (sofia/external/16502928610 at 10.191.210.148) Locked, Waiting on external entities 2013-01-25 18:21:22.150301 [NOTICE] switch_core_session.c:1498 Session 64 (sofia/external/16502928610 at 10.191.210.148) Ended 2013-01-25 18:21:22.150301 [NOTICE] switch_core_session.c:1502 Close Channel sofia/external/16502928610 at 10.191.210.148 [CS_DESTROY] 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:554 (sofia/external/16502928610 at 10.191.210.148) Callstate Change HANGUP -> DOWN 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:557 (sofia/external/16502928610 at 10.191.210.148) Running State Change CS_DESTROY 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:567 (sofia/external/16502928610 at 10.191.210.148) State DESTROY 2013-01-25 18:21:22.150301 [DEBUG] mod_sofia.c:394 sofia/external/ 16502928610 at 10.191.210.148 SOFIA DESTROY 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:99 sofia/external/16502928610 at 10.191.210.148 Standard DESTROY 2013-01-25 18:21:22.150301 [DEBUG] switch_core_state_machine.c:567 (sofia/external/16502928610 at 10.191.210.148) State DESTROY going to sleep freeswitch at internal> On Fri, Jan 25, 2013 at 5:42 PM, Michael Collins wrote: > Well, I did think of one thing that you could try. If it works it would be > kind of cute: > > Answer the call like you currently do, grab the digits from the caller > Set the transfer_ringback variable to ${hold_music} (or whatever you want > them to hear) > Transfer the call to another extension, like this: > > > Create a new extension like this: > > > > > > > > ... do whatever you need to do with the call after the data dip > > > > The theory is that the "please hold while we look up your info..." sound > is the "transfer ringback" sound. The target extension "data_dip" then > "rings" while it's looking up the info and then "answers" after the lookup > is complete. > > I didn't try it so please remember that the standard disclaimerapplies. Let us know if you make any headway on this. > > -MC > > > On Fri, Jan 25, 2013 at 4:32 PM, JP wrote: > >> MC, >> Thanks for the info. Unfortunately outbound event socket application is >> not an option for me right now. I was wondering if the following "ideas" >> can be of any use to me( I know I am trying clutch at straws, but hope is >> eternal :)) >> 1. Use "bgapi". Although on the wiki I only see this used with >> "originate". >> 2. Can I use a conference to park the caller there while playing music on >> another call leg of the conference. Meanwhile can I execute the Lua script >> :)) >> 3. Can I launch another Lua script from my original one asynchronously? >> 4. Can I have a start up Lua script running in the background listening >> for some custom events and I can have the main caller thread generate the >> custom event and then start playing long music file. Once the background >> Lua script is done with the data dip, it can just break the music on the >> other channel. >> >> That's it from me. Someone help me from sinking... please. >> >> Thanks, >> -JP >> >> >> On Fri, Jan 25, 2013 at 4:10 PM, Michael Collins wrote: >> >>> The problem you're running into is that you are trying to do two >>> different things inside of the thread that is controlling the call flow. >>> Playing sound files and running Lua scripts are synchronous in this >>> context. This seems more suited for an outbound event socket application. >>> Using the event socket you can control the call while doing other things. >>> >>> Just curious - does anyone have an ESL example of this kind of thing? >>> The wiki could definitely use this kind of example as it is quite useful. >>> I'll look around and see if I can find anything. >>> >>> -MC >>> >>> >>> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >>> >>>> Avi, >>>> Thanks for the response. Unfortunately that did not work. This is what >>>> I did >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> The music only plays after the Lua script exits. I played around with >>>> the timer value, but that did not help either. Any other ideas? >>>> >>>> Thanks >>>> JP >>>> >>>> >>>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>>> >>>>> ESL gives you more control, especially to execute things in parallel. >>>>> If you want to do it within FS, that's a little more complicated, >>>>> since most things are sequential. >>>>> >>>>> However, if you use the async APIs, then I think it could work: >>>>> 1) sched_broadcast -- >>>>> schedule the playback of your file on the UUID of the channel, for +0, >>>>> meaning now. >>>>> 2) run your lua script >>>>> 3) Then stop (break) the playback on that uuid: break >>>>> 4) Continue with your execution. >>>>> >>>>> Please wikify your results and let us know how that works out! >>>>> -Avi >>>>> >>>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>>> >>>>>> I want to do the following... >>>>>> >>>>>> 1. Play a long music file from a dialplan. >>>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>>> some data dip. >>>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>>> music and continue in the dialplan >>>>>> >>>>>> How do I do this. Any sample code or pointers would be greatly >>>>>> appreciated. >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/8ad5885e/attachment-0001.html From freeswitch at nullinfinity.org Sat Jan 26 05:59:41 2013 From: freeswitch at nullinfinity.org (Johan Brannlund) Date: Fri, 25 Jan 2013 21:59:41 -0500 Subject: [Freeswitch-users] PSTN via Google Talk Message-ID: <20130125215941.0503d8a0@mars> Sanjay Soni wrote: > Have you specified audio codec in autoload_config/dingaling.conf.xml > as ? I thought I had, but a look revealed that I hadn't. I put that in, and now everything is working perfectly. Many thanks! - Johan From brian.wiese.freeswitch at gmail.com Sat Jan 26 06:46:08 2013 From: brian.wiese.freeswitch at gmail.com (Brian Wiese) Date: Fri, 25 Jan 2013 21:46:08 -0600 Subject: [Freeswitch-users] Executing multiple applications in parallel In-Reply-To: References: Message-ID: JP: Give this a try: ===== Dialplan Extension ===== ========== ===== Lua Script (wait.lua) ===== api = freeswitch.API(); uuid = argv[1]; --The UUID of the call. if uuid == nil then freeswitch.consoleLog("crit","Fatal Error: A UUID was not passed to the script!\n"); else freeswitch.consoleLog("info","Working...\n"); --Do the task here that may take some time... session = freeswitch.Session(uuid); freeswitch.consoleLog("info","Transferring call!\n"); session:transfer("lua_done"); --Obviously set this to a dialplan extension... end ========== Using this method you can even make the dialplan as one extension (use extension "29000step2" for session:transfer in the Lua script): ==== Dialplan Single Extension Example ===== ========== ~Brian On Thu, Jan 24, 2013 at 7:34 PM, JP wrote: > I want to do the following... > > 1. Play a long music file from a dialplan. > 2. While the music is playing, I want to launch a Lua script to do some data > dip. > 3. Once the data is fetched and Lua exits, I want to interrupt the music and > continue in the dialplan > > How do I do this. Any sample code or pointers would be greatly appreciated. > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From itsusama at gmail.com Sat Jan 26 07:26:31 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Sat, 26 Jan 2013 09:26:31 +0500 Subject: [Freeswitch-users] Executing Multiple Applications in parallel In-Reply-To: References: Message-ID: http://pastebin.freeswitch.org/20508 I just checked the link and it works... On Sat, Jan 26, 2013 at 5:44 AM, Usama Zaidi wrote: > Hey, > > I posted this earlier as a question, but I think this might be relevant. > ttp://pastebin.freeswitch.org/20508 . > The workaround (hack) for my issue is to check the connection state > ($con->connected()) every time you loop and reconnect and subscribe for > events again when that returns false. > > On Sat, Jan 26, 2013 at 5:33 AM, < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: Executing multiple applications in parallel (Michael Collins) >> 2. Re: Executing multiple applications in parallel (JP) >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins >> To: FreeSWITCH Users Help >> Cc: >> Date: Fri, 25 Jan 2013 16:10:37 -0800 >> Subject: Re: [Freeswitch-users] Executing multiple applications in >> parallel >> The problem you're running into is that you are trying to do two >> different things inside of the thread that is controlling the call flow. >> Playing sound files and running Lua scripts are synchronous in this >> context. This seems more suited for an outbound event socket application. >> Using the event socket you can control the call while doing other things. >> >> Just curious - does anyone have an ESL example of this kind of thing? The >> wiki could definitely use this kind of example as it is quite useful. I'll >> look around and see if I can find anything. >> >> -MC >> >> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >> >>> Avi, >>> Thanks for the response. Unfortunately that did not work. This is what I >>> did >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> The music only plays after the Lua script exits. I played around with >>> the timer value, but that did not help either. Any other ideas? >>> >>> Thanks >>> JP >>> >>> >>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>> >>>> ESL gives you more control, especially to execute things in parallel. >>>> If you want to do it within FS, that's a little more complicated, since >>>> most things are sequential. >>>> >>>> However, if you use the async APIs, then I think it could work: >>>> 1) sched_broadcast -- >>>> schedule the playback of your file on the UUID of the channel, for +0, >>>> meaning now. >>>> 2) run your lua script >>>> 3) Then stop (break) the playback on that uuid: break >>>> 4) Continue with your execution. >>>> >>>> Please wikify your results and let us know how that works out! >>>> -Avi >>>> >>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>> >>>>> I want to do the following... >>>>> >>>>> 1. Play a long music file from a dialplan. >>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>> some data dip. >>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>> music and continue in the dialplan >>>>> >>>>> How do I do this. Any sample code or pointers would be greatly >>>>> appreciated. >>>>> >>>>> Thanks >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> ---------- Forwarded message ---------- >> From: JP >> To: FreeSWITCH Users Help >> Cc: >> Date: Fri, 25 Jan 2013 16:32:50 -0800 >> Subject: Re: [Freeswitch-users] Executing multiple applications in >> parallel >> MC, >> Thanks for the info. Unfortunately outbound event socket application is >> not an option for me right now. I was wondering if the following "ideas" >> can be of any use to me( I know I am trying clutch at straws, but hope is >> eternal :)) >> 1. Use "bgapi". Although on the wiki I only see this used with >> "originate". >> 2. Can I use a conference to park the caller there while playing music on >> another call leg of the conference. Meanwhile can I execute the Lua script >> :)) >> 3. Can I launch another Lua script from my original one asynchronously? >> 4. Can I have a start up Lua script running in the background listening >> for some custom events and I can have the main caller thread generate the >> custom event and then start playing long music file. Once the background >> Lua script is done with the data dip, it can just break the music on the >> other channel. >> >> That's it from me. Someone help me from sinking... please. >> >> Thanks, >> -JP >> >> >> On Fri, Jan 25, 2013 at 4:10 PM, Michael Collins wrote: >> >>> The problem you're running into is that you are trying to do two >>> different things inside of the thread that is controlling the call flow. >>> Playing sound files and running Lua scripts are synchronous in this >>> context. This seems more suited for an outbound event socket application. >>> Using the event socket you can control the call while doing other things. >>> >>> Just curious - does anyone have an ESL example of this kind of thing? >>> The wiki could definitely use this kind of example as it is quite useful. >>> I'll look around and see if I can find anything. >>> >>> -MC >>> >>> >>> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >>> >>>> Avi, >>>> Thanks for the response. Unfortunately that did not work. This is what >>>> I did >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> The music only plays after the Lua script exits. I played around with >>>> the timer value, but that did not help either. Any other ideas? >>>> >>>> Thanks >>>> JP >>>> >>>> >>>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>>> >>>>> ESL gives you more control, especially to execute things in parallel. >>>>> If you want to do it within FS, that's a little more complicated, >>>>> since most things are sequential. >>>>> >>>>> However, if you use the async APIs, then I think it could work: >>>>> 1) sched_broadcast -- >>>>> schedule the playback of your file on the UUID of the channel, for +0, >>>>> meaning now. >>>>> 2) run your lua script >>>>> 3) Then stop (break) the playback on that uuid: break >>>>> 4) Continue with your execution. >>>>> >>>>> Please wikify your results and let us know how that works out! >>>>> -Avi >>>>> >>>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>>> >>>>>> I want to do the following... >>>>>> >>>>>> 1. Play a long music file from a dialplan. >>>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>>> some data dip. >>>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>>> music and continue in the dialplan >>>>>> >>>>>> How do I do this. Any sample code or pointers would be greatly >>>>>> appreciated. >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > I'd love to change the world, but they wont gimme the source code to it > -- I'd love to change the world, but they wont gimme the source code to it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130126/2b82d2b6/attachment-0001.html From anthony.minessale at gmail.com Sat Jan 26 07:37:44 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 25 Jan 2013 22:37:44 -0600 Subject: [Freeswitch-users] Executing Multiple Applications in parallel In-Reply-To: References: Message-ID: only non-ESL method to do what you want.... 1) make the dialplan use the FSAPI via variable expansion to call luarun on the script with the uuid as the first arg (this will go in a new thread) 2) play your long file 3) do what you need to do and then use the uuid to feed to uuid_transfer api On Fri, Jan 25, 2013 at 10:26 PM, Usama Zaidi wrote: > http://pastebin.freeswitch.org/20508 I just checked the link and it > works... > > > On Sat, Jan 26, 2013 at 5:44 AM, Usama Zaidi wrote: > >> Hey, >> >> I posted this earlier as a question, but I think this might be relevant. >> ttp://pastebin.freeswitch.org/20508. >> The workaround (hack) for my issue is to check the connection state >> ($con->connected()) every time you loop and reconnect and subscribe for >> events again when that returns false. >> >> On Sat, Jan 26, 2013 at 5:33 AM, < >> freeswitch-users-request at lists.freeswitch.org> wrote: >> >>> Send FreeSWITCH-users mailing list submissions to >>> freeswitch-users at lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request at lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner at lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of FreeSWITCH-users digest..." >>> >>> Today's Topics: >>> >>> 1. Re: Executing multiple applications in parallel (Michael Collins) >>> 2. Re: Executing multiple applications in parallel (JP) >>> >>> >>> ---------- Forwarded message ---------- >>> From: Michael Collins >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Fri, 25 Jan 2013 16:10:37 -0800 >>> Subject: Re: [Freeswitch-users] Executing multiple applications in >>> parallel >>> The problem you're running into is that you are trying to do two >>> different things inside of the thread that is controlling the call flow. >>> Playing sound files and running Lua scripts are synchronous in this >>> context. This seems more suited for an outbound event socket application. >>> Using the event socket you can control the call while doing other things. >>> >>> Just curious - does anyone have an ESL example of this kind of thing? >>> The wiki could definitely use this kind of example as it is quite useful. >>> I'll look around and see if I can find anything. >>> >>> -MC >>> >>> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >>> >>>> Avi, >>>> Thanks for the response. Unfortunately that did not work. This is what >>>> I did >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> The music only plays after the Lua script exits. I played around with >>>> the timer value, but that did not help either. Any other ideas? >>>> >>>> Thanks >>>> JP >>>> >>>> >>>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>>> >>>>> ESL gives you more control, especially to execute things in parallel. >>>>> If you want to do it within FS, that's a little more complicated, >>>>> since most things are sequential. >>>>> >>>>> However, if you use the async APIs, then I think it could work: >>>>> 1) sched_broadcast -- >>>>> schedule the playback of your file on the UUID of the channel, for +0, >>>>> meaning now. >>>>> 2) run your lua script >>>>> 3) Then stop (break) the playback on that uuid: break >>>>> 4) Continue with your execution. >>>>> >>>>> Please wikify your results and let us know how that works out! >>>>> -Avi >>>>> >>>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>>> >>>>>> I want to do the following... >>>>>> >>>>>> 1. Play a long music file from a dialplan. >>>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>>> some data dip. >>>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>>> music and continue in the dialplan >>>>>> >>>>>> How do I do this. Any sample code or pointers would be greatly >>>>>> appreciated. >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: JP >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Fri, 25 Jan 2013 16:32:50 -0800 >>> Subject: Re: [Freeswitch-users] Executing multiple applications in >>> parallel >>> MC, >>> Thanks for the info. Unfortunately outbound event socket application is >>> not an option for me right now. I was wondering if the following "ideas" >>> can be of any use to me( I know I am trying clutch at straws, but hope is >>> eternal :)) >>> 1. Use "bgapi". Although on the wiki I only see this used with >>> "originate". >>> 2. Can I use a conference to park the caller there while playing music >>> on another call leg of the conference. Meanwhile can I execute the Lua >>> script :)) >>> 3. Can I launch another Lua script from my original one asynchronously? >>> 4. Can I have a start up Lua script running in the background listening >>> for some custom events and I can have the main caller thread generate the >>> custom event and then start playing long music file. Once the background >>> Lua script is done with the data dip, it can just break the music on the >>> other channel. >>> >>> That's it from me. Someone help me from sinking... please. >>> >>> Thanks, >>> -JP >>> >>> >>> On Fri, Jan 25, 2013 at 4:10 PM, Michael Collins wrote: >>> >>>> The problem you're running into is that you are trying to do two >>>> different things inside of the thread that is controlling the call flow. >>>> Playing sound files and running Lua scripts are synchronous in this >>>> context. This seems more suited for an outbound event socket application. >>>> Using the event socket you can control the call while doing other things. >>>> >>>> Just curious - does anyone have an ESL example of this kind of thing? >>>> The wiki could definitely use this kind of example as it is quite useful. >>>> I'll look around and see if I can find anything. >>>> >>>> -MC >>>> >>>> >>>> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >>>> >>>>> Avi, >>>>> Thanks for the response. Unfortunately that did not work. This is what >>>>> I did >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> The music only plays after the Lua script exits. I played around with >>>>> the timer value, but that did not help either. Any other ideas? >>>>> >>>>> Thanks >>>>> JP >>>>> >>>>> >>>>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>>>> >>>>>> ESL gives you more control, especially to execute things in parallel. >>>>>> If you want to do it within FS, that's a little more complicated, >>>>>> since most things are sequential. >>>>>> >>>>>> However, if you use the async APIs, then I think it could work: >>>>>> 1) sched_broadcast -- >>>>>> schedule the playback of your file on the UUID of the channel, for +0, >>>>>> meaning now. >>>>>> 2) run your lua script >>>>>> 3) Then stop (break) the playback on that uuid: break >>>>>> 4) Continue with your execution. >>>>>> >>>>>> Please wikify your results and let us know how that works out! >>>>>> -Avi >>>>>> >>>>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>>>> >>>>>>> I want to do the following... >>>>>>> >>>>>>> 1. Play a long music file from a dialplan. >>>>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>>>> some data dip. >>>>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>>>> music and continue in the dialplan >>>>>>> >>>>>>> How do I do this. Any sample code or pointers would be greatly >>>>>>> appreciated. >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> I'd love to change the world, but they wont gimme the source code to it >> > > > > -- > I'd love to change the world, but they wont gimme the source code to it > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130125/55cb2987/attachment-0001.html From ben at langfeld.co.uk Sat Jan 26 14:21:23 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Sat, 26 Jan 2013 11:21:23 +0000 Subject: [Freeswitch-users] Executing multiple applications in parallel In-Reply-To: References: Message-ID: For anyone wondering, Adhearsion (http://adhearsion.com) can do this kind of thing quite easily. I know you said ES is not an option for you, but you might consider it in future. Regards, Ben Langfeld On 26 January 2013 03:46, Brian Wiese wrote: > JP: > > Give this a try: > > ===== Dialplan Extension ===== > > > > > data="ivr/ivr-on_hold_indefinitely.wav"/> > > > > > > > > > ========== > > ===== Lua Script (wait.lua) ===== > api = freeswitch.API(); > > uuid = argv[1]; --The UUID of the call. > > if uuid == nil then > freeswitch.consoleLog("crit","Fatal Error: A UUID was not passed to > the script!\n"); > else > freeswitch.consoleLog("info","Working...\n"); > > --Do the task here that may take some time... > > session = freeswitch.Session(uuid); > freeswitch.consoleLog("info","Transferring call!\n"); > session:transfer("lua_done"); --Obviously set this to a dialplan > extension... > end > ========== > > > Using this method you can even make the dialplan as one extension (use > extension "29000step2" for session:transfer in the Lua script): > > ==== Dialplan Single Extension Example ===== > > > break="on-true"> > > > data="ivr/ivr-on_hold_indefinitely.wav"/> > > > > > > > > > > > > data="ivr/ivr-yes_we_have_no_bananas.wav"/> > > > > > ========== > > ~Brian > > On Thu, Jan 24, 2013 at 7:34 PM, JP wrote: > > I want to do the following... > > > > 1. Play a long music file from a dialplan. > > 2. While the music is playing, I want to launch a Lua script to do some > data > > dip. > > 3. Once the data is fetched and Lua exits, I want to interrupt the music > and > > continue in the dialplan > > > > How do I do this. Any sample code or pointers would be greatly > appreciated. > > > > Thanks > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130126/7b5fa47c/attachment.html From avi at avimarcus.net Sat Jan 26 19:42:07 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 26 Jan 2013 18:42:07 +0200 Subject: [Freeswitch-users] Caller-ID variables and another Caller-ID question In-Reply-To: References: Message-ID: > > The next question is about the outgoing Caller-ID Name. From the SIP >> trace, I see the name is set both in the "From:" line and the >> "Remote-Party-ID:" line, in quotes (""). However, the name does not appear >> on the caller-ID box of the POTS line. It seems the phone company (AT&T) >> is getting the name from a CNAM lookup. Is this something I can change, or >> is it just not sent in the path between Flowroute and AT&T? >> > If the phone company is doing a CNAM lookup then there's not really > anything you can do, unless they have a reason that they're doing a CNAM > lookup instead of using the caller_id_name information. I'd ask Flowroute > for feedback on that one. > Just to comment on this: The norm in the USA is when calls hit the POTS network, there is no name attached. The receiving carrier needs to do a CNAM lookup. In that case, to have the name show up properly, you need to register the CNAM name. The exception is: some carriers have direct connections and they skip the POTS system, so they can use the actual SIP Name, as sent. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130126/8d78aa7c/attachment.html From vgoget at yahoo.com Sat Jan 26 22:26:32 2013 From: vgoget at yahoo.com (VG Oget) Date: Sat, 26 Jan 2013 11:26:32 -0800 (PST) Subject: [Freeswitch-users] Please ignore previous messages from this account - got hacked In-Reply-To: <1359209583.59834.BPMail_high_noncarrier@web122005.mail.ne1.yahoo.com> References: <1359209583.59834.BPMail_high_noncarrier@web122005.mail.ne1.yahoo.com> Message-ID: <1359228392.30271.YahooMailNeo@web122004.mail.ne1.yahoo.com> From fs-list at communicatefreely.net Sun Jan 27 04:43:30 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Sat, 26 Jan 2013 20:43:30 -0500 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE after 28 minutes of talking? Message-ID: <51048642.4070303@communicatefreely.net> Hello, I'm getting frequent complaints from a customer about calls being dropped. It seems to be specific to this customer, as opposed to system wide. I found a call that they mentioned, and the call hangs up on the phone side (as opposed to the trunk side), with NORMAL_TEMPORARY_FAILURE after about 28 minutes of successful media. What would cause a call to cut off and return this cause code? Is there any way that loss of Internet connectivity to the endpoint would do this, or do we have to receive some kind of SIP message to generate that? Any help is appreciated. -Tim From victor.chukalovskiy at gmail.com Sun Jan 27 04:54:01 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Sat, 26 Jan 2013 20:54:01 -0500 Subject: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE after 28 minutes of talking? In-Reply-To: <51048642.4070303@communicatefreely.net> References: <51048642.4070303@communicatefreely.net> Message-ID: <510488B9.7050703@gmail.com> Hi Tim, I suspect either side might be dropping on a timer (session-expire). Tha would happen if customer's phone does not support or does not honour session-timers. Cheers, Victor On 01/26/2013 08:43 PM, Tim St. Pierre wrote: > Hello, > > I'm getting frequent complaints from a customer about calls being > dropped. It seems to be specific to this customer, as opposed to system > wide. I found a call that they mentioned, and the call hangs up on the > phone side (as opposed to the trunk side), with NORMAL_TEMPORARY_FAILURE > after about 28 minutes of successful media. What would cause a call to > cut off and return this cause code? Is there any way that loss of > Internet connectivity to the endpoint would do this, or do we have to > receive some kind of SIP message to generate that? > > Any help is appreciated. > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jaykris at gmail.com Sun Jan 27 05:45:44 2013 From: jaykris at gmail.com (JP) Date: Sat, 26 Jan 2013 18:45:44 -0800 Subject: [Freeswitch-users] Executing Multiple Applications in parallel In-Reply-To: References: Message-ID: Thanks Anthony!! That works perfectly. This is what I did... This is pretty neat. It should be on the wiki. If someone can suggest the best place to place this on the wiki, I can update it. Thank you, JP On Fri, Jan 25, 2013 at 8:37 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > only non-ESL method to do what you want.... > > 1) make the dialplan use the FSAPI via variable expansion to call luarun > on the script with the uuid as the first arg (this will go in a new thread) > 2) play your long file > 3) do what you need to do and then use the uuid to feed to uuid_transfer > api > > > > > On Fri, Jan 25, 2013 at 10:26 PM, Usama Zaidi wrote: > >> http://pastebin.freeswitch.org/20508 I just checked the link and it >> works... >> >> >> On Sat, Jan 26, 2013 at 5:44 AM, Usama Zaidi wrote: >> >>> Hey, >>> >>> I posted this earlier as a question, but I think this might be relevant. >>> ttp://pastebin.freeswitch.org/20508. >>> The workaround (hack) for my issue is to check the connection state >>> ($con->connected()) every time you loop and reconnect and subscribe for >>> events again when that returns false. >>> >>> On Sat, Jan 26, 2013 at 5:33 AM, < >>> freeswitch-users-request at lists.freeswitch.org> wrote: >>> >>>> Send FreeSWITCH-users mailing list submissions to >>>> freeswitch-users at lists.freeswitch.org >>>> >>>> To subscribe or unsubscribe via the World Wide Web, visit >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> or, via email, send a message with subject or body 'help' to >>>> freeswitch-users-request at lists.freeswitch.org >>>> >>>> You can reach the person managing the list at >>>> freeswitch-users-owner at lists.freeswitch.org >>>> >>>> When replying, please edit your Subject line so it is more specific >>>> than "Re: Contents of FreeSWITCH-users digest..." >>>> >>>> Today's Topics: >>>> >>>> 1. Re: Executing multiple applications in parallel (Michael Collins) >>>> 2. Re: Executing multiple applications in parallel (JP) >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Michael Collins >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Fri, 25 Jan 2013 16:10:37 -0800 >>>> Subject: Re: [Freeswitch-users] Executing multiple applications in >>>> parallel >>>> The problem you're running into is that you are trying to do two >>>> different things inside of the thread that is controlling the call flow. >>>> Playing sound files and running Lua scripts are synchronous in this >>>> context. This seems more suited for an outbound event socket application. >>>> Using the event socket you can control the call while doing other things. >>>> >>>> Just curious - does anyone have an ESL example of this kind of thing? >>>> The wiki could definitely use this kind of example as it is quite useful. >>>> I'll look around and see if I can find anything. >>>> >>>> -MC >>>> >>>> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >>>> >>>>> Avi, >>>>> Thanks for the response. Unfortunately that did not work. This is what >>>>> I did >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> The music only plays after the Lua script exits. I played around with >>>>> the timer value, but that did not help either. Any other ideas? >>>>> >>>>> Thanks >>>>> JP >>>>> >>>>> >>>>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>>>> >>>>>> ESL gives you more control, especially to execute things in parallel. >>>>>> If you want to do it within FS, that's a little more complicated, >>>>>> since most things are sequential. >>>>>> >>>>>> However, if you use the async APIs, then I think it could work: >>>>>> 1) sched_broadcast -- >>>>>> schedule the playback of your file on the UUID of the channel, for +0, >>>>>> meaning now. >>>>>> 2) run your lua script >>>>>> 3) Then stop (break) the playback on that uuid: break >>>>>> 4) Continue with your execution. >>>>>> >>>>>> Please wikify your results and let us know how that works out! >>>>>> -Avi >>>>>> >>>>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>>>> >>>>>>> I want to do the following... >>>>>>> >>>>>>> 1. Play a long music file from a dialplan. >>>>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>>>> some data dip. >>>>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>>>> music and continue in the dialplan >>>>>>> >>>>>>> How do I do this. Any sample code or pointers would be greatly >>>>>>> appreciated. >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: JP >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Fri, 25 Jan 2013 16:32:50 -0800 >>>> Subject: Re: [Freeswitch-users] Executing multiple applications in >>>> parallel >>>> MC, >>>> Thanks for the info. Unfortunately outbound event socket application is >>>> not an option for me right now. I was wondering if the following "ideas" >>>> can be of any use to me( I know I am trying clutch at straws, but hope is >>>> eternal :)) >>>> 1. Use "bgapi". Although on the wiki I only see this used with >>>> "originate". >>>> 2. Can I use a conference to park the caller there while playing music >>>> on another call leg of the conference. Meanwhile can I execute the Lua >>>> script :)) >>>> 3. Can I launch another Lua script from my original one asynchronously? >>>> 4. Can I have a start up Lua script running in the background listening >>>> for some custom events and I can have the main caller thread generate the >>>> custom event and then start playing long music file. Once the background >>>> Lua script is done with the data dip, it can just break the music on the >>>> other channel. >>>> >>>> That's it from me. Someone help me from sinking... please. >>>> >>>> Thanks, >>>> -JP >>>> >>>> >>>> On Fri, Jan 25, 2013 at 4:10 PM, Michael Collins wrote: >>>> >>>>> The problem you're running into is that you are trying to do two >>>>> different things inside of the thread that is controlling the call flow. >>>>> Playing sound files and running Lua scripts are synchronous in this >>>>> context. This seems more suited for an outbound event socket application. >>>>> Using the event socket you can control the call while doing other things. >>>>> >>>>> Just curious - does anyone have an ESL example of this kind of thing? >>>>> The wiki could definitely use this kind of example as it is quite useful. >>>>> I'll look around and see if I can find anything. >>>>> >>>>> -MC >>>>> >>>>> >>>>> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >>>>> >>>>>> Avi, >>>>>> Thanks for the response. Unfortunately that did not work. This is >>>>>> what I did >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> The music only plays after the Lua script exits. I played around with >>>>>> the timer value, but that did not help either. Any other ideas? >>>>>> >>>>>> Thanks >>>>>> JP >>>>>> >>>>>> >>>>>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>>>>> >>>>>>> ESL gives you more control, especially to execute things in parallel. >>>>>>> If you want to do it within FS, that's a little more complicated, >>>>>>> since most things are sequential. >>>>>>> >>>>>>> However, if you use the async APIs, then I think it could work: >>>>>>> 1) sched_broadcast -- >>>>>>> schedule the playback of your file on the UUID of the channel, for +0, >>>>>>> meaning now. >>>>>>> 2) run your lua script >>>>>>> 3) Then stop (break) the playback on that uuid: break >>>>>>> 4) Continue with your execution. >>>>>>> >>>>>>> Please wikify your results and let us know how that works out! >>>>>>> -Avi >>>>>>> >>>>>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>>>>> >>>>>>>> I want to do the following... >>>>>>>> >>>>>>>> 1. Play a long music file from a dialplan. >>>>>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>>>>> some data dip. >>>>>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>>>>> music and continue in the dialplan >>>>>>>> >>>>>>>> How do I do this. Any sample code or pointers would be greatly >>>>>>>> appreciated. >>>>>>>> >>>>>>>> Thanks >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Michael S Collins >>>>> Twitter: @mercutioviz >>>>> http://www.FreeSWITCH.org >>>>> http://www.ClueCon.com >>>>> http://www.OSTAG.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> I'd love to change the world, but they wont gimme the source code to it >>> >> >> >> >> -- >> I'd love to change the world, but they wont gimme the source code to it >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130126/a5267282/attachment-0001.html From shaheryarkh at gmail.com Sun Jan 27 20:43:41 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Sun, 27 Jan 2013 18:43:41 +0100 Subject: [Freeswitch-users] FreeSWITCH website down Message-ID: Hi, It seems that http://freeswitch.org as well as http://wiki.freeswitch.orgare down or at least unaccessible. Is it a scheduled maintenance or yet another DDOS? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130127/bc89576f/attachment.html From krice at freeswitch.org Sun Jan 27 21:09:20 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 27 Jan 2013 12:09:20 -0600 Subject: [Freeswitch-users] FreeSWITCH website down In-Reply-To: Message-ID: Its back up an running... No DDoS this time... Just a hung webserver... On 1/27/13 11:43 AM, "Muhammad Shahzad" wrote: > Hi, > > > It seems that http://freeswitch.org as well as http://wiki.freeswitch.org are > down or at least unaccessible. Is it a scheduled maintenance or yet another > DDOS? > > Thank you. > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130127/ebd53fa4/attachment.html From gamar at center.com Sun Jan 27 23:24:47 2013 From: gamar at center.com (Gilbert Amar) Date: Sun, 27 Jan 2013 12:24:47 -0800 Subject: [Freeswitch-users] building mod_opal and v1.2stable In-Reply-To: References: <000701cdf4f4$cee11ef0$6ca35cd0$@center.com> <4A8B8CFC-2B43-45F1-AD6E-99ADA1E9DC75@gmail.com> <004101cdfa60$25b1b4e0$71151ea0$@center.com> Message-ID: <009201cdfccc$5af22900$10d67b00$@center.com> Hello, I did fill two Jira bugs, one for mod_323 and another for mod_opal. Just wanted to remind that since those two do not work, at least for me, there is no more any support in FS 1.2 stable for H323. So I ask again did someone succeeded in having any of those 2 working with 1.2 stable on a Linux platform ? Gilbert From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, January 25, 2013 12:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] building mod_opal and v1.2stable Gilbert, Thank you for collecting all of this information. You did quite a bit of work and that will help the developers figure out what's going on. Would you mind creating a ticket at jira.freeswitch.org and adding all this information? That will help the developers keep track of the issue and avoid any unnecessary delays. Thanks! -MC On Thu, Jan 24, 2013 at 10:25 AM, Gilbert Amar wrote: Hello, I succeeded in building mod_opal, but it crash FreeSWITCH (segmentation default) after simple test. Here are the steps on Debian "squeeze". After getting all required packages including swig Advice, check that you do not have any version of ptlib or h323plus or opal already somewhere. use find /- name "*libpt*.so" use find /- name "*libopal*.so" My FreeSWITCH is in /usr/local/src/ and I am root when I build. cd /usr/local/src/ svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_12_0/ ptlib-2.12.0 cd /usr/local/src/ptlib-2.12.0 ./configure --disable-plugins --disable-v4l2 --prefix=/opt/lib make && make install export PKG_CONFIG_PATH=/opt/lib/lib/pkgconfig cd /usr/local/src/ svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/tags/v3_12_0 opal-3.12.0 cd /usr/local/src/opal-3.12.0 ./configure --disable-plugins --prefix=/opt/lib make && make install As the install fails to copy include/ep/ cd /usr/local/src/opal-3.12.0 mkdir -p /opt/lib/include/opal/ep ; chmod 755 /opt/lib/include/opal/ep ; ( for fn in include/ep/*.h ; do /usr/bin/install -c -m 644 $fn /opt/lib/include/opal/ep ; done); Please notice that the mkdir line is one long line. Add mod_opal to your modules.conf Go to your mod_opal src dir in my case cd /usr/local/src/freeswitch/src/mod/endpoints/mod_opal Edit mod_opal.h You need to change 2 lines #include to #include then #define HAVE_T38 (OPAL_CHECK_VERSION(3,11,2) && OPAL_T38_CAPABILITY) to #define HAVE_T38 0 Now you can build mod_opal type make Go back to your FS dir and do make and make install to check all is ok. Before launch FreeSWITCH you need to set LD_LIBRARY_PATH export LD_LIBRARY_PATH=/opt/lib/lib start your FreeSWITCH Then try to call extension 9197 it should work despite the errors lines like 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not initialized! 2013-01-24 03:12:25.597058 [WARNING] switch_core_codec.c:802 Codec is not initialized! 2013-01-24 03:12:25.597058 [ERR] switch_core_timer.c:117 Timer is not properly configured. In my case I have a sip phone registered to FS at extension 1000 If I call from openphone this extension the call get thru, I pick up but mod_opal isn't aware of this and when I hang up on the sip phone FreeSWITCH crash. Segmentation fault /opt/freeswitch-v1.2stable/bin/freeswitch -conf /home/center/freeswitch/conf -log /home/center/freeswitch/log -db /home/center/freeswitch/db -sounds /home/center/freeswitch/sounds -nonat -nonatmap Gilbert From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Wednesday, January 23, 2013 3:26 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] building mod_opal and v1.2stable Do you mean you built opal in /root/opal? You shouldn't build things as root, but that's not the reason for your troubles. I have to say I'm a bit puzzled, it looks like you've installed opal to multiple locations. /usr/local/freeswitch/opal looks the best match (opal/localep.h) but it's in the wrong location. /usr/local/src/freeswitch/libs/opal/include/ep/localep.h Not sure why it's installed here, opal isn't shipped under libs/ /usr/local/include/ep/localep.h /usr/include/ep/localep.h Close but the directory is wrong as it needs to be opal/localep.h, not ep/localep.h /usr/include/opal/localep.h Should be the correct filename, but you don't have a copy. It could be a change in the opal library but I find that unlikely since it'd break apps using the library like FS. More likely there's a problem with the prefixes given to opal's configure command. -Steve On 23 January 2013 10:46, Anton Rostotskiy wrote: Search results: /usr/local/freeswitch/opal/include/opal/localep.h /usr/local/src/freeswitch/libs/opal/include/ep/localep.h /usr/local/include/ep/localep.h /usr/include/ep/localep.h /root/opal/include/ep/localep.h As I said earlier, I installed ptlib and opal in /root directory. 2013/1/23 Steven Ayre It's localep.h, not local.h: /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.h:41:26: error: opal/localep.h: No such file or directory find / -name localep.h Sorry, seems my phone did an annoying autouncorrection! -Steve On 23 January 2013 08:16, Anton Rostotskiy wrote: Thanks for reply, Steven! Tried to search local.h file, but search returned nothing. That are all local.h files in my system: /usr/local/src/freeswitch/libs/spandsp/src/gsm0610_local.h /usr/local/src/freeswitch/libs/spandsp/src/t30_local.h /usr/local/src/freeswitch/libs/spandsp/src/t42_t43_local.h About -devel packages - at first tried to install ptlib, opal and -devel from yum but their versions have been too low for mod_opal. 2013/1/22 Steven Ayre Run: find / -name local.h It's failing to include opal/local.h. All the other errors are because its missing the definitions in that file. Normally I would say to instal the -devel package, but since you built from svn that doesn't apply. Perhaps it's not installed or installed to the wrong place. The find command will tell us where that file is. Steve on iPhone On 22 Jan 2013, at 16:07, Anton Rostotskiy wrote: Also tried to install mod_opal and also failed. CentOS 6.3 x86_64 ptlib and opal are the last version obtained from svn as it is said in wiki . but just a bunch of errors after make mod_opal http://pastebin.com/HHUPLCLV 2013/1/18 Steven Ayre The latest version of opal in Debian (even in Sid) isn't new enough for mod_opal. It needed various features added that are only only available in the svn version. The wiki page states you have to install ptlib and opal from svn - this is why. http://wiki.freeswitch.org/wiki/Mod_opal Of course that can then cause conflicts unless you package it yourself or install a dummy package, or problems with other libraries/apps that are built against the libopal that's already packaged. Ptlib and h323plus/opal are also notorious for having to get matching versions. If you don't pair exactly the right version of ptlib with the specific version of h323plus/opal you can get instability problems. So the official Debian ptlib package probably wouldn't be good with opal from svn... As for mod_h323, it does compile and work, at least on Wheezy and possibly Squeeze too. I've never had any luck getting mod_opal working but have got mod_h323 working. I don't actively use it though - all my traffic is currently SIP. Another option is to run Yate as a SIP->H323 proxy. It only handles the signalling, media still goes directly caller->FS. It worked quite well, although it would occasionally crash - but at least that only dropped the H323 calls and didn't take FS down with it, and Yate automatically restarted. That was on Lenny so a long time ago... Steve on iPhone On 17 Jan 2013, at 20:54, "Gilbert Amar" wrote: Hi, I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and opal 3.10.7 on a debian Ptlib and opal get build with no error, but when trying mod_opal I get Compiling /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DP_64BIT -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, switch_frame_t**, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: error: 'class OpalCall' has no member named 'IsSwitchingT38' /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, const switch_frame_t*, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: error: 'class OpalCall' has no member named 'IsSwitchingT38' make[3]: *** [mod_opal.lo] Error 1 make[2]: *** [install] Error 1 make[1]: *** [mod_opal-install] Error 1 make: *** [mod_opal-install] Error 2 Has anyone succeded in building mod_opal ? Should I go with mod_h323 instead ? Thanks for your help _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- ? ?????????, ????? ????????? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- ? ?????????, ????? ????????? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- ? ?????????, ????? ????????? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130127/841b71c8/attachment-0001.html From johnthan123 at gmail.com Sun Jan 27 16:16:31 2013 From: johnthan123 at gmail.com (johnthan123) Date: Sun, 27 Jan 2013 18:46:31 +0530 Subject: [Freeswitch-users] Need Help On mod_xml_curl Message-ID: HI All, Can You Please any one help me To Write Directory and Dialplan and In XML_CURL Like Below #################CODE######################## #!/usr/bin/perl use strict; use warnings; use Data::Dump qw(dump); use CGI; my $q = CGI->new; my $section = $q->param('section'); if ($section eq 'directory'){ $XML_STRING2 = '
'; } elsif ( $section eq 'dialplan' ) { $XML_STRING = '
'; } ###########################END############################# This is Working for Me But i cant able to take the Database values into XML file Thanks In Advance Thanks J :: T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130127/7b2416f2/attachment.html From rnbrady at gmail.com Mon Jan 28 02:02:25 2013 From: rnbrady at gmail.com (Richard Brady) Date: Sun, 27 Jan 2013 23:02:25 +0000 Subject: [Freeswitch-users] adding In-Reply-To in sip header not working In-Reply-To: References: Message-ID: Ok, nifty. They are letting you present a number you do not own as Caller ID on an outbound call if that outbound call is a forwarded leg of an inbound call. They do this by looking the In-Reply-To header of the INVITE for the forwarded leg, which should contain the Call-ID of the orignal leg. So you need to copy the *Call-ID* in order to authorize the *Caller ID*. A couple things: 1. From the docs: *effective_caller_id_name Sets the effective callerid name. This is automatically exported to the B-leg; however, it is not valid in an origination string. In other words, set this before calling bridge, otherwise use origination_caller_id_name* 2. You shouldn't care about 1 above as it should be copied across from the A leg by default and you are not modifying it, so remove effective_caller_id_name and don't bother with origination_caller_id_name either. 3. You should use sip_h_ not sip_rh_ because you want the header in the new INVITE going out. Perhaps try: Hope this helps. Richard On 21 January 2013 19:54, Mario G wrote: > Thanks, apparently I had it wrong, the doc below states that the PBX must > support it incoming, they pointed me to using effective_caller_id which I > added to the bridge but it still does not work. Would love to fix this > since the cell phones currently have no idea who is calling. > Mario G > > "{originate_timeout=45,alert_info=n=${lua_ringtone}}${group_call(bria@ > ${domain_name}+A)},${group_call(deskphone@ > ${domain_name}+A)},[leg_delay_start=20,leg_timeout=23,effective_caller_id_number=${caller_id_number}]sofia/gateway/${dial_gateway}/19161234567" > /> > > > - > > Please note that this feature is ONLY AVAILABLE for customers using a > SIP PBX that either supports (or allows the configuring of) the > "in-reply-to" header (defined by RFC 3261) for incoming calls which are > forwarded to an outbound trunk. In these instances Callcentric will > "Pass-Through" the CallerID from the original call which was received to > the outbound bridged/forwarded call. > > On Jan 19, 2013, at 5:13 PM, Richard Brady wrote: > > On 20 January 2013 00:06, Mario G wrote: > >> I never did this so I must be missing something, I tried both below but >> the bridge then fails. Can anyone shed some light on what I am doing wrong. >> My ITSP now supports in-reply-to so I can pass the caller ID to a forwarded >> call from FS. >> > > In-Reply-To should contain a Call-ID not a caller ID. They are very > different. > > The following would make a bit more sense, but still not a lot: > > > > Using In-Reply-To in a response doesn't seem right to me. I would expect > it to appear in an INVITE. For for example, you get a missed call and you > call the person back, then the INVITE for the callback would have a new > Call-ID but the original Call-ID in the In-Reply-To header. That said, I > have no idea what your ITSPs intended use for the header is. > > Richard > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130127/bdd55db8/attachment.html From shaheryarkh at gmail.com Mon Jan 28 03:52:59 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 28 Jan 2013 01:52:59 +0100 Subject: [Freeswitch-users] How to auto save voicemail message? Message-ID: Hi, When a caller leaves a voicemail for callee using FS, an IVR is played to the caller to press e.g. 1 to save the message, press 2 to listen to it ... and so on. Is there any voicemail module parameter / variable we can set in dialplan to skip this prompt entirely, simply save the message and hangup if caller hasn't already hanged up. Typically if somebody calls me on my cell and i am not available to pickup the phone, call automatically sent to my voicemail box by my operator, where caller can record his message and just hangup, without having to wait for the IVR, give input to it and then hangup. Is the same behaviour possible in FS voicemail module? Please help. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/688aedff/attachment.html From avi at avimarcus.net Mon Jan 28 10:30:13 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 28 Jan 2013 09:30:13 +0200 Subject: [Freeswitch-users] How to auto save voicemail message? In-Reply-To: References: Message-ID: If you call FS's voicemail, record a message, and hang up -- it gets saved. That's the behavior users expect. -Avi On Mon, Jan 28, 2013 at 2:52 AM, Muhammad Shahzad wrote: > Hi, > > When a caller leaves a voicemail for callee using FS, an IVR is played to > the caller to press e.g. 1 to save the message, press 2 to listen to it ... > and so on. Is there any voicemail module parameter / variable we can set in > dialplan to skip this prompt entirely, simply save the message and hangup > if caller hasn't already hanged up. > > Typically if somebody calls me on my cell and i am not available to pickup > the phone, call automatically sent to my voicemail box by my operator, > where caller can record his message and just hangup, without having to wait > for the IVR, give input to it and then hangup. Is the same behaviour > possible in FS voicemail module? Please help. > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/28bd427a/attachment-0001.html From alessio at asistar.it Mon Jan 28 10:45:32 2013 From: alessio at asistar.it (Alessio) Date: Mon, 28 Jan 2013 08:45:32 +0100 Subject: [Freeswitch-users] Patton FXO configuration on Freeswitch - more details In-Reply-To: References: <5102AAF6.2070201@asistar.it> Message-ID: <51062C9C.8020701@asistar.it> Nella citazione in data venerd? 25 gennaio 2013 20:17:36, Michael Collins ha scritto: > Alessio, > > That "error" is just explaining to you that the caller does not have > an entry in the XML directory, therefore there's no way to perform > authentication. You'll need to create a user with the ID mentioned in > the error message and then set the password in there as well as in the > Patton. (I've never done a Patton set up but I've done several other > analog gateway devices.) > > Hope this helps. > > -Michael > > On Fri, Jan 25, 2013 at 7:55 AM, Alessio > wrote: > > Hi all! > > We're trying to configure a Patton FXO in our Freeswitch. > > We have an internal PBX > - Freeswitch 1.3.6 installed from svn master with (demo) setup > on IP > 10.0.1.190 > - Patton 4112 with simple config on IP 10.0.1.10 > > Find more info below. > So any help would be great! > > We set it up that the Patton registers with the PBX and we get the > error: > > --------------------- > 2013-01-25 16:44:39.433873 [WARNING] sofia_reg.c:2469 Can't find user > [fxo0_3_0 at 10.0.1.190 ] > You must define a domain called '10.0.1.190' in your directory and > add a > user with the id="fxo0_3_0" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > --------------------- > > We defined a gateway in conf/sip_profiles/external/fxo.xml: > --------------------- > > --> > > > > > > --------------------- > > And the relevant parts of the Patton config: > --------------------- > # define auth > authentication-service authenticate > username fxo0_3_0 password gtatFAePDQFL0btl3cbsaA== encrypted > > # patton registers > location-service LOCATION_FXO_3_0 > domain 1 10.0.1.190 5060 > identity-group group_fxo0_3_0 > authentication outbound > authenticate 1 authentication-service authenticate username > fxo0_3_0 > identity fxo0_3_0 > authentication outbound > authenticate 1 authentication-service authenticate username > fxo0_3_0 > > registration outbound > registrar 10.0.1.190 5060 > lifetime 180 > register auto > retry-timeout on-system-error 10 > retry-timeout on-client-error 10 > retry-timeout on-server-error 10 > > call outbound > use profile tone-set IT > use profile voip default > use profile sip default > preferred-transport-protocol tcp > traffic-class local-default > invite-transaction-timeout 32 > non-invite-transaction-timeout 32 > penalty-box > > call inbound > use profile tone-set IT > use profile voip default > use profile sip default > > context sip-gateway ASISTAR_3_0 > interface fxo0_3_0 > bind interface eth0 context router port 5060 > > context sip-gateway ASISTAR_3_0 > bind location-service LOCATION_FXO_3_0 > no shutdown > > --------------------- > Output from 'sofia status' > --------------------- > > external::10.0.1.10 gateway sip:fxo0_3_0 at 10.0.1.10 > NOREG > > > > Thanks, > Alessio > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Hi Michael, thanks for the reply. So if I register in directory as a phone how do I get out of the gateway, for example through a bridge? Alessio From madan.mallikarjun at outlook.com Mon Jan 28 12:20:17 2013 From: madan.mallikarjun at outlook.com (madan.mallikarjun) Date: Mon, 28 Jan 2013 01:20:17 -0800 (PST) Subject: [Freeswitch-users] Music on Hold Message-ID: <1359364817390-7586777.post@n2.nabble.com> HI all, how do we change the music on hold for a particular group ie assuming i have 2 groups for which i do not want to play the default MOH for 1 of the groups thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Music-on-Hold-tp7586777.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Mon Jan 28 12:26:12 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 28 Jan 2013 03:26:12 -0600 Subject: [Freeswitch-users] Music on Hold In-Reply-To: <1359364817390-7586777.post@n2.nabble.com> Message-ID: See http://wiki.freeswitch.org/wiki/Variable_hold_music On 1/28/13 3:20 AM, "madan.mallikarjun" wrote: > HI all, > > how do we change the music on hold for a particular group ie assuming i have > 2 groups for which i do not want to play the default MOH for 1 of the groups > > > thanks > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Music-on-Hold-tp7586777.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From stuart.mills3 at btopenworld.com Mon Jan 28 12:26:59 2013 From: stuart.mills3 at btopenworld.com (Stuart Mills) Date: Mon, 28 Jan 2013 09:26:59 -0000 Subject: [Freeswitch-users] softphones on Amazon Message-ID: Hi, I have a strange problem using FreeSwitch on an Amazon EC2 instance and was wondering if anyone out there could help me. The problem seems to be that I can register an extension from a remote network in say France, phone works ok but when passing calls to it (from sip peer through JavaScript to user extension) I get one way audio, the caller can hear me on the softphone but I can?t hear them ? like the reverse audio path can?t be established. That would be ok and fixable if the issue was consistent but on another phone registered from within the UK, it all works fine and I get a good 2 way conversation. If I dial from one softphone in France directly to the other in the UK, that works too. I followed the wiki?s guide to installing on Amazon and have everything right, otherwise I doubt it would work at all. Is anyone able to help me or have experience of this type of issue? Maybe it?s a local router issue to the phone in France but I didn?t need to do anything special to the one in the UK to work, so not sure it could be that either. Kind Regards, Stuart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/18a20ad6/attachment.html From madan.mallikarjun at outlook.com Mon Jan 28 12:37:35 2013 From: madan.mallikarjun at outlook.com (madan.mallikarjun) Date: Mon, 28 Jan 2013 01:37:35 -0800 (PST) Subject: [Freeswitch-users] Music on Hold In-Reply-To: References: <1359364817390-7586777.post@n2.nabble.com> Message-ID: hi this actually changes in the dial plan but i need to change it only for a group. thanks Date: Mon, 28 Jan 2013 01:34:55 -0800 From: ml-node+s2379917n7586778h31 at n2.nabble.com To: madan.mallikarjun at outlook.com Subject: Re: Music on Hold See http://wiki.freeswitch.org/wiki/Variable_hold_music On 1/28/13 3:20 AM, "madan.mallikarjun" <[hidden email]> wrote: > HI all, > > how do we change the music on hold for a particular group ie assuming i have > 2 groups for which i do not want to play the default MOH for 1 of the groups > > > thanks > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Music-on-Hold-tp7586777.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > [hidden email] > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: [hidden email] http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org If you reply to this email, your message will be added to the discussion below: http://freeswitch-users.2379917.n2.nabble.com/Music-on-Hold-tp7586777p7586778.html To unsubscribe from Music on Hold, click here. NAML -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Music-on-Hold-tp7586777p7586780.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/3690b2b7/attachment-0001.html From shaheryarkh at gmail.com Mon Jan 28 13:07:03 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 28 Jan 2013 11:07:03 +0100 Subject: [Freeswitch-users] How to auto save voicemail message? In-Reply-To: References: Message-ID: For some reason its not working for me. I have FS behind OpenSIPS, serving only voicemail, so when call comes in on FS, i simply send it to voicemail directly. Here is my dial plan. But its not working accordingly. Do you suspect anything wrong in it? Which might be causing it? Thank you. On Mon, Jan 28, 2013 at 8:30 AM, Avi Marcus wrote: > If you call FS's voicemail, record a message, and hang up -- it gets > saved. That's the behavior users expect. > -Avi > > > On Mon, Jan 28, 2013 at 2:52 AM, Muhammad Shahzad wrote: > >> Hi, >> >> When a caller leaves a voicemail for callee using FS, an IVR is played to >> the caller to press e.g. 1 to save the message, press 2 to listen to it ... >> and so on. Is there any voicemail module parameter / variable we can set in >> dialplan to skip this prompt entirely, simply save the message and hangup >> if caller hasn't already hanged up. >> >> Typically if somebody calls me on my cell and i am not available to >> pickup the phone, call automatically sent to my voicemail box by my >> operator, where caller can record his message and just hangup, without >> having to wait for the IVR, give input to it and then hangup. Is the same >> behaviour possible in FS voicemail module? Please help. >> >> Thank you. >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 <+49%20176%2099%2083%2010%2085> >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/84621068/attachment.html From andrew at cassidywebservices.co.uk Mon Jan 28 13:08:22 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 28 Jan 2013 10:08:22 +0000 Subject: [Freeswitch-users] softphones on Amazon In-Reply-To: References: Message-ID: One way audio is generally a NAT/Firewall issue at the client end, and probably not related to EC2 at all. There has been a lot of work in the wiki recently regarding NAT issues. On 28 January 2013 09:26, Stuart Mills wrote: > Hi, > > I have a strange problem using FreeSwitch on an Amazon EC2 instance and > was wondering if anyone out there could help me. > > The problem seems to be that I can register an extension from a remote > network in say France, phone works ok but when passing calls to it (from > sip peer through JavaScript to user extension) I get one way audio, the > caller can hear me on the softphone but I can?t hear them ? like the > reverse audio path can?t be established. > > That would be ok and fixable if the issue was consistent but on another > phone registered from within the UK, it all works fine and I get a good 2 > way conversation. > > If I dial from one softphone in France directly to the other in the UK, > that works too. > > I followed the wiki?s guide to installing on Amazon and have everything > right, otherwise I doubt it would work at all. > > Is anyone able to help me or have experience of this type of issue? Maybe > it?s a local router issue to the phone in France but I didn?t need to do > anything special to the one in the UK to work, so not sure it could be that > either. > > Kind Regards, > > Stuart > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/c6c6afeb/attachment.html From shaheryarkh at gmail.com Mon Jan 28 13:09:05 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 28 Jan 2013 11:09:05 +0100 Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH website down In-Reply-To: <5105b052.c6ea440a.79fd.4a0f@mx.google.com> References: <5105b052.c6ea440a.79fd.4a0f@mx.google.com> Message-ID: Yes, it was down for a couple of hours yesterday. Then Ken checked and fix web server, which was hang. Thanks guys for checking it from different regions of the world. :-) Thank you. On Sun, Jan 27, 2013 at 11:55 PM, mercutio.viz at gmail.com < mercutio.viz at gmail.com> wrote: > I just accessed them both, so possibly a temp outage. > > Sent from my HTC EVO 4G LTE exclusively from Sprint > > > ----- Reply message ----- > From: "Muhammad Shahzad" > To: "FreeSWITCH Users Help" , < > freeswitch-dev at lists.freeswitch.org> > Subject: [Freeswitch-dev] FreeSWITCH website down > Date: Sun, Jan 27, 2013 9:43 AM > > > Hi, > > > It seems that http://freeswitch.org as well as http://wiki.freeswitch.orgare down or at least unaccessible. Is it a scheduled maintenance or yet > another DDOS? > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/9a538a0d/attachment-0001.html From steveayre at gmail.com Mon Jan 28 13:12:34 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 28 Jan 2013 10:12:34 +0000 Subject: [Freeswitch-users] softphones on Amazon In-Reply-To: References: Message-ID: One way audio usually is a result of NAT. Not all NAT routers operate in the same way, and not all softphones handle it in the same way, so that may explain why it works in one location but not the other. I assume the softphones were behind NAT? Set the FS log level to debug, and check the IP in the remote SDP. You need the phone's external IP in there. Phones usually determine that using STUN. Some NAT routers have a feature called SIP ALG to rewrite it for phones not doing it themselves, but that often will conflict with phones that have already put the correct IP (and also cannot work with TLS). In short check where RTP is being sent to is correct... Without the correct IP FS also has a last-resort feature where it detects the IP the caller sends audio from and replies to there (RTP auto-adjust). A log entry indicates when that happens. But that requires FS to receive audio, which means it can't work until the caller starts speaking after the call is answered (so might miss hearing the callee say 'hello'), and the caller will never hear ringing (unless generated locally by the phone when it gets the SIP 183/180 response). The NAT page on the Wiki may be informative: http://wiki.freeswitch.org/wiki/NAT Just to complicate things Amazon EC2 runs behind a NAT layer too AFAIK. I don't have any personal experience with EC2. -Steve On 28 January 2013 09:26, Stuart Mills wrote: > Hi, > > I have a strange problem using FreeSwitch on an Amazon EC2 instance and > was wondering if anyone out there could help me. > > The problem seems to be that I can register an extension from a remote > network in say France, phone works ok but when passing calls to it (from > sip peer through JavaScript to user extension) I get one way audio, the > caller can hear me on the softphone but I can?t hear them ? like the > reverse audio path can?t be established. > > That would be ok and fixable if the issue was consistent but on another > phone registered from within the UK, it all works fine and I get a good 2 > way conversation. > > If I dial from one softphone in France directly to the other in the UK, > that works too. > > I followed the wiki?s guide to installing on Amazon and have everything > right, otherwise I doubt it would work at all. > > Is anyone able to help me or have experience of this type of issue? Maybe > it?s a local router issue to the phone in France but I didn?t need to do > anything special to the one in the UK to work, so not sure it could be that > either. > > Kind Regards, > > Stuart > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/fc2ccdb0/attachment.html From jaganthoutam at gmail.com Mon Jan 28 14:06:14 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Mon, 28 Jan 2013 16:36:14 +0530 Subject: [Freeswitch-users] Need Help On mod_xml_curl In-Reply-To: References: Message-ID: Hi all Any Update On 27 January 2013 18:46, johnthan123 wrote: > HI All, > > > Can You Please any one help me To Write Directory and Dialplan and In > XML_CURL Like Below > > #################CODE######################## > #!/usr/bin/perl > use strict; > use warnings; > use Data::Dump qw(dump); > use CGI; > my $q = CGI->new; > my $section = $q->param('section'); > > if ($section eq 'directory'){ > > $XML_STRING2 = ' > > >
> > > > > > > > > > > > > > > >
> '; > } > elsif ( $section eq 'dialplan' ) { > $XML_STRING = ' >
> > > > > > > > > >
> > '; > } > > ###########################END############################# > > This is Working for Me But i cant able to take the Database values into XML > file > > > Thanks In Advance > > > Thanks > > J :: T > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From avi at avimarcus.net Mon Jan 28 14:22:26 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 28 Jan 2013 13:22:26 +0200 Subject: [Freeswitch-users] How to auto save voicemail message? In-Reply-To: References: Message-ID: Get a log in FS of the call. Maybe it's a permission issue. The logs should help. -Avi (This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.) On Jan 28, 2013 12:10 PM, "Muhammad Shahzad" wrote: For some reason its not working for me. I have FS behind OpenSIPS, serving only voicemail, so when call comes in on FS, i simply send it to voicemail directly. Here is my dial plan. But its not working accordingly. Do you suspect anything wrong in it? Which might be causing it? Thank you. On Mon, Jan 28, 2013 at 8:30 AM, Avi Marcus wrote: > > If you call FS's voice... _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/4bffb4ec/attachment.html From stuart.mills3 at btopenworld.com Mon Jan 28 15:38:33 2013 From: stuart.mills3 at btopenworld.com (Stuart Mills) Date: Mon, 28 Jan 2013 12:38:33 -0000 Subject: [Freeswitch-users] softphones on Amazon In-Reply-To: References: Message-ID: Cheers Steven, I guessed it?s something to do with NAT ? most likely at the softphone end. I?m using X-Lite in all softphone cases, so it?s literally the end user?s PC and router that?s different. I tried STUN but that seemed to make things worse where no audio was heard from both ends. I?ll run some debug and check the remote SDP to see what it contains and go from there. Stuart From: Steven Ayre Sent: Monday, January 28, 2013 10:12 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] softphones on Amazon One way audio usually is a result of NAT. Not all NAT routers operate in the same way, and not all softphones handle it in the same way, so that may explain why it works in one location but not the other. I assume the softphones were behind NAT? Set the FS log level to debug, and check the IP in the remote SDP. You need the phone's external IP in there. Phones usually determine that using STUN. Some NAT routers have a feature called SIP ALG to rewrite it for phones not doing it themselves, but that often will conflict with phones that have already put the correct IP (and also cannot work with TLS). In short check where RTP is being sent to is correct... Without the correct IP FS also has a last-resort feature where it detects the IP the caller sends audio from and replies to there (RTP auto-adjust). A log entry indicates when that happens. But that requires FS to receive audio, which means it can't work until the caller starts speaking after the call is answered (so might miss hearing the callee say 'hello'), and the caller will never hear ringing (unless generated locally by the phone when it gets the SIP 183/180 response). The NAT page on the Wiki may be informative: http://wiki.freeswitch.org/wiki/NAT Just to complicate things Amazon EC2 runs behind a NAT layer too AFAIK. I don't have any personal experience with EC2. -Steve On 28 January 2013 09:26, Stuart Mills wrote: Hi, I have a strange problem using FreeSwitch on an Amazon EC2 instance and was wondering if anyone out there could help me. The problem seems to be that I can register an extension from a remote network in say France, phone works ok but when passing calls to it (from sip peer through JavaScript to user extension) I get one way audio, the caller can hear me on the softphone but I can?t hear them ? like the reverse audio path can?t be established. That would be ok and fixable if the issue was consistent but on another phone registered from within the UK, it all works fine and I get a good 2 way conversation. If I dial from one softphone in France directly to the other in the UK, that works too. I followed the wiki?s guide to installing on Amazon and have everything right, otherwise I doubt it would work at all. Is anyone able to help me or have experience of this type of issue? Maybe it?s a local router issue to the phone in France but I didn?t need to do anything special to the one in the UK to work, so not sure it could be that either. Kind Regards, Stuart _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/8dd410e2/attachment-0001.html From gvvsubhashkumar at gmail.com Mon Jan 28 15:45:38 2013 From: gvvsubhashkumar at gmail.com (Subhash) Date: Mon, 28 Jan 2013 18:15:38 +0530 Subject: [Freeswitch-users] Dequeue DTMF from FreeTDM Channel In-Reply-To: References: Message-ID: Moy, Any update on this? Thanks, Subhash. On Jan 24, 2013 12:30 PM, "Subhash" wrote: > Moy, > > Did you get time to look into this issue? > > Thanks, > Subhash. > On Tue, Jan 22, 2013 at 6:19 PM, Subhash wrote: > >> Ok. >> We did the test again,pasted the freeswitch log and uploaded the >> wireshark of the both success and failed call traces.Please let me know if >> you have problem in accessing the URLs . >> >> http://pastebin.freeswitch.org/20484 >> >> http://www.cloudshark.org/captures/6dc99dc2f68c -- Success >> >> http://www.cloudshark.org/captures/27b88b2c8ca4--- Failure >> >> >> >> >> Thanks, >> Subhash. >> >> >> On Tue, Jan 22, 2013 at 12:06 AM, Michael Collins wrote: >> >>> That may be the case, but that doesn't actually supply the information >>> you need to find out where the DTMFs are coming from. I would recommend >>> doing the test that Moy suggested. At least find out where FreeSWITCH >>> thinks this DTMF is coming from. >>> >>> -MC >>> >>> >>> On Sun, Jan 20, 2013 at 6:43 PM, Subhash wrote: >>> >>>> But we don't see any problem when both endpoints are SIP. >>>> >>>> Thanks, >>>> Subhash. >>>> >>>> >>>> On Mon, Jan 21, 2013 at 7:12 AM, Moises Silva wrote: >>>> >>>>> On Fri, Jan 18, 2013 at 1:27 AM, Subhash wrote: >>>>> >>>>>> Yes,we are trying to bridge PSTN and VOIP call. >>>>>> >>>>>> We posted the log in pastebin.Please use the below URL to access the >>>>>> freeswitch log. >>>>>> >>>>>> http://pastebin.freeswitch.org/20461 >>>>>> >>>>>> >>>>> >>>>> >>>>> 1. 013-01-16 01:02:26.954165 [INFO] >>>>> ftmod_sangoma_isdn_stack_rcv.c:144 [s4c1][4:1] Received >>>>> CONNECT/CONNECT ACK (suId:1 suInstId:2 spInstId:2 ces:0) >>>>> 2. 2013-01-16 01:02:26.954165 [DEBUG] >>>>> ftmod_sangoma_isdn_stack_hndl.c:285 [s4c1][4:1] Processing >>>>> CONNECT/CONNECT ACK (suId:1 suInstId:2 spInstId:2 ces:0) >>>>> 3. 2013-01-16 01:02:26.974165 [DEBUG] switch_rtp.c:3596 Correct >>>>> ip/port confirmed. >>>>> 4. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:1002 [s4c1] >>>>> [4:1] First packet read stats: Rx queue len: 4, Rx queue size: 10 >>>>> 5. 2013-01-16 01:02:26.974165 [DEBUG] ftmod_wanpipe.c:938 [s4c1][ >>>>> 4:1] First packet write stats: Tx queue len: 1, Tx queue size: 5, >>>>> Tx idle: 10 >>>>> 6. 2013-01-16 01:02:27.694163 [DEBUG] switch_rtp.c:3797 RTP RECV >>>>> DTMF 9:640 >>>>> 7. 2013-01-16 01:02:27.694163 [DEBUG] switch_ivr_bridge.c:393 Send >>>>> signal FreeTDM/4:1/2001 [BREAK] >>>>> 8. 2013-01-16 01:02:27.694163 [DEBUG] ftdm_io.c:3979 [s4c1][4:1] Generating >>>>> DTMF [9] >>>>> >>>>> >>>>> The DTMF comes from the RTP side, not FreeTDM (as shown by the log >>>>> above). You have to verify using tcpdump/wireshark if there is any real >>>>> DTMF being received. >>>>> >>>>> I'd do exactly the same test again, but this time I would also start a >>>>> wireshark trace. Then please pastebin the trace again and upload the pcap >>>>> trace to cloudshark (http://www.cloudshark.org/) and paste the link >>>>> here. >>>>> >>>>> Moy >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/580e1ff9/attachment.html From roger.castaldo at gmail.com Mon Jan 28 16:48:51 2013 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Mon, 28 Jan 2013 08:48:51 -0500 Subject: [Freeswitch-users] Need Help On mod_xml_curl In-Reply-To: References: Message-ID: Please refer to the link http://bit.ly/VKwFGY On Mon, Jan 28, 2013 at 6:06 AM, Jagadish Thoutam wrote: > Hi all > > > Any Update > > On 27 January 2013 18:46, johnthan123 wrote: > > HI All, > > > > > > Can You Please any one help me To Write Directory and Dialplan and In > > XML_CURL Like Below > > > > #################CODE######################## > > #!/usr/bin/perl > > use strict; > > use warnings; > > use Data::Dump qw(dump); > > use CGI; > > my $q = CGI->new; > > my $section = $q->param('section'); > > > > if ($section eq 'directory'){ > > > > $XML_STRING2 = ' > > > > > >
> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >
> > '; > > } > > elsif ( $section eq 'dialplan' ) { > > $XML_STRING = ' > >
> > > > > > > > > > > > data="sofia/gateway/peer_asterisk/$1"/> > > > > > > > >
> > > > '; > > } > > > > ###########################END############################# > > > > This is Working for Me But i cant able to take the Database values into > XML > > file > > > > > > Thanks In Advance > > > > > > Thanks > > > > J :: T > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/46fa1877/attachment-0001.html From steveayre at gmail.com Mon Jan 28 16:49:09 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 28 Jan 2013 13:49:09 +0000 Subject: [Freeswitch-users] Need Help On mod_xml_curl In-Reply-To: References: Message-ID: You need to return valid XML, your dialplan code path doesn't open the document tag. I suggest you read the Wiki page on the module, it gives several examples of the response you need to generate. http://wiki.freeswitch.org/wiki/Mod_xml_curl#XML_Usage -Steve On 27 January 2013 13:16, johnthan123 wrote: > > HI All, > > > Can You Please any one help me To Write Directory and Dialplan and In XML_CURL Like Below > > #################CODE######################## > #!/usr/bin/perl > use strict; > use warnings; > use Data::Dump qw(dump); > use CGI; > my $q = CGI->new; > my $section = $q->param('section'); > > if ($section eq 'directory'){ > > $XML_STRING2 = ' > > >
> > > > > > > > > > > > > > > >
> '; > } > elsif ( $section eq 'dialplan' ) { > $XML_STRING = ' >
> > > > > > > > > >
> > '; > } > > ###########################END############################# > > This is Working for Me But i cant able to take the Database values into XML file > > > Thanks In Advance > > > Thanks > > J :: T > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/c27879b2/attachment.html From steveayre at gmail.com Mon Jan 28 16:50:32 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 28 Jan 2013 13:50:32 +0000 Subject: [Freeswitch-users] Need Help On mod_xml_curl In-Reply-To: References: Message-ID: Once the response is nested in the correct document, section etc tags the XML syntax is identical to the static dialplan/directory/etc local files, except that you only need to return the parts relevant to the call. -Steve On 28 January 2013 13:49, Steven Ayre wrote: > You need to return valid XML, your dialplan code path doesn't open the document > tag. > > I suggest you read the Wiki page on the module, it gives several examples > of the response you need to generate. > > http://wiki.freeswitch.org/wiki/Mod_xml_curl#XML_Usage > > -Steve > > > > On 27 January 2013 13:16, johnthan123 wrote: > > > > HI All, > > > > > > Can You Please any one help me To Write Directory and Dialplan and In > XML_CURL Like Below > > > > #################CODE######################## > > #!/usr/bin/perl > > use strict; > > use warnings; > > use Data::Dump qw(dump); > > use CGI; > > my $q = CGI->new; > > my $section = $q->param('section'); > > > > if ($section eq 'directory'){ > > > > $XML_STRING2 = ' > > > > > >
> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >
> > '; > > } > > elsif ( $section eq 'dialplan' ) { > > $XML_STRING = ' > >
> > > > > > > > > > > > data="sofia/gateway/peer_asterisk/$1"/> > > > > > > > >
> > > > '; > > } > > > > ###########################END############################# > > > > This is Working for Me But i cant able to take the Database values into > XML file > > > > > > Thanks In Advance > > > > > > Thanks > > > > J :: T > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/53d5d11f/attachment.html From miconda at gmail.com Mon Jan 28 19:33:52 2013 From: miconda at gmail.com (Daniel-Constantin Mierla) Date: Mon, 28 Jan 2013 17:33:52 +0100 Subject: [Freeswitch-users] [OT] Kamailio World Conference Message-ID: <5106A870.9060404@gmail.com> Hello, I would like to announce Kamailio World Conference, event to take place in Berlin, Germany, during April 15-16, 2013. More details can be found at: - http://conference.kamailio.com The event targets to be a broad VoIP and RTC professionals gathering in Europe, not focusing only on Kamailio. There are presentations from other projects, including some using FreeSwitch. Call for speakers is still open, I would like to invite more participants that want to give a presentations in the area of RTC. Last, but not least, many thanks to FreeSwitch core developers and Cluecon organizers for supporting the event, Cluecon being one of the models we try to follow, blending technical presentations with business aspects. I hope to meet many of you in Berlin, if not, maybe later this year in Chicago at Cluecon. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com - From mehroz.ashraf85 at gmail.com Mon Jan 28 15:10:27 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Mon, 28 Jan 2013 04:10:27 -0800 (PST) Subject: [Freeswitch-users] Direcotry Status with mod_xml_curl Message-ID: <1359375027857-7586787.post@n2.nabble.com> Hi, I have been working with asterisk previously, recently came over Freeswitch platform. I dig up for the similar fucntionality of asteriks-realtime and found MOD_XML_CURL as the nearest match for dynamic directory(extension) creation. One good thing about asterisk-realtime was, it was compact in a single table and direct conversation with Asterisk and DB is possible , whereas in Freeswitch its through Webserver and curl requests. Watever, i accepted that! but one more thing i am missing is the status of SIP Registration (registered or not). Again, in asterisk realtime i can see the RTP port assigned to extension when it registers, this way i can see the current status reading the DB , and port number. Now the question is , how can i see this with mod_xml_curl ? and also, if any one can help me with the above mentioned grudges and crushes! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Direcotry-Status-with-mod-xml-curl-tp7586787.html Sent from the freeswitch-users mailing list archive at Nabble.com. From sanjay.sanjaysoni at gmail.com Mon Jan 28 16:57:38 2013 From: sanjay.sanjaysoni at gmail.com (soni4m) Date: Mon, 28 Jan 2013 05:57:38 -0800 (PST) Subject: [Freeswitch-users] Webrtc In-Reply-To: References: Message-ID: <1359381458946-7586790.post@n2.nabble.com> Hi Anthony, Can you please explain how the overall solution will look like once this webrtc support in FS effort completes ? Will it be the way asterisk is used in many of these demos (One example here ) In a sense that sipml5 (SIP stack in the browser) is necessarily there on the browser side. My interest here is to understand the topology here how FS will help connecting the legacy SIP devices with browser and what will come into the picture in between. Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Webrtc-tp7580875p7586790.html Sent from the freeswitch-users mailing list archive at Nabble.com. From cal.leeming at simplicitymedialtd.co.uk Mon Jan 28 20:07:57 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 28 Jan 2013 17:07:57 +0000 Subject: [Freeswitch-users] Direcotry Status with mod_xml_curl In-Reply-To: <1359375027857-7586787.post@n2.nabble.com> References: <1359375027857-7586787.post@n2.nabble.com> Message-ID: Hello, You want to use mod_event_socket to retrieve the registration status of a user. See the following; http://wiki.freeswitch.org/wiki/Event_Socket Hope this helps. Cal On Mon, Jan 28, 2013 at 12:10 PM, mehroz wrote: > Hi, > > I have been working with asterisk previously, recently came over Freeswitch > platform. > > I dig up for the similar fucntionality of asteriks-realtime and found > MOD_XML_CURL as the nearest match for dynamic directory(extension) > creation. > One good thing about asterisk-realtime was, it was compact in a single > table > and direct conversation with Asterisk and DB is possible , whereas in > Freeswitch its through Webserver and curl requests. Watever, i accepted > that! but one more thing i am missing is the status of SIP Registration > (registered or not). > > Again, in asterisk realtime i can see the RTP port assigned to extension > when it registers, this way i can see the current status reading the DB , > and port number. > > Now the question is , how can i see this with mod_xml_curl ? and also, if > any one can help me with the above mentioned grudges and crushes! > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Direcotry-Status-with-mod-xml-curl-tp7586787.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/abb2ff9d/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon Jan 28 20:10:23 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 28 Jan 2013 17:10:23 +0000 Subject: [Freeswitch-users] Need Help On mod_xml_curl In-Reply-To: References: Message-ID: Hi Jonathan, On a separate side note, that code is particularly unclean. You may want to consider either building the XML response using an XML library, or use a template parsing language so you are not placing raw XML in your source. In regards to your original email, the link Steven has provided is a good starting point. Hope this helps. Cal On Sun, Jan 27, 2013 at 1:16 PM, johnthan123 wrote: > HI All, > > > Can You Please any one help me To Write Directory and Dialplan and In > XML_CURL Like Below > > #################CODE######################## > #!/usr/bin/perl > use strict; > use warnings; > use Data::Dump qw(dump); > use CGI; > my $q = CGI->new; > my $section = $q->param('section'); > > if ($section eq 'directory'){ > > $XML_STRING2 = ' > > >
> > > > > > > > > > > > > > > >
> '; > } > elsif ( $section eq 'dialplan' ) { > $XML_STRING = ' >
> > > > > > data="sofia/gateway/peer_asterisk/$1"/> > > > >
> > '; > } > > ###########################END############################# > > This is Working for Me But i cant able to take the Database values into > XML file > > > Thanks In Advance > > > Thanks > > J :: T > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/4435071f/attachment.html From steveayre at gmail.com Mon Jan 28 20:45:36 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 28 Jan 2013 17:45:36 +0000 Subject: [Freeswitch-users] Direcotry Status with mod_xml_curl In-Reply-To: References: <1359375027857-7586787.post@n2.nabble.com> Message-ID: It's also possible to store the information in a ODBC database (core-db-dsn) and mod_sofia's odbc-dsn profile param. You can then query (read-only!) the registration DB tables directory. If you want to interact in realtime (ie no polling) collecting events and running async api commands via ESL is the way to go. It's similar to Asterisk's AMI and the events are very powerful. -Steve On 28 January 2013 17:07, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello, > > You want to use mod_event_socket to retrieve the registration status of a > user. > > See the following; > http://wiki.freeswitch.org/wiki/Event_Socket > > Hope this helps. > > Cal > > > On Mon, Jan 28, 2013 at 12:10 PM, mehroz wrote: > >> Hi, >> >> I have been working with asterisk previously, recently came over >> Freeswitch >> platform. >> >> I dig up for the similar fucntionality of asteriks-realtime and found >> MOD_XML_CURL as the nearest match for dynamic directory(extension) >> creation. >> One good thing about asterisk-realtime was, it was compact in a single >> table >> and direct conversation with Asterisk and DB is possible , whereas in >> Freeswitch its through Webserver and curl requests. Watever, i accepted >> that! but one more thing i am missing is the status of SIP Registration >> (registered or not). >> >> Again, in asterisk realtime i can see the RTP port assigned to extension >> when it registers, this way i can see the current status reading the DB , >> and port number. >> >> Now the question is , how can i see this with mod_xml_curl ? and also, if >> any one can help me with the above mentioned grudges and crushes! >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Direcotry-Status-with-mod-xml-curl-tp7586787.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/0ab0be48/attachment-0001.html From msc at freeswitch.org Mon Jan 28 20:56:33 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Jan 2013 09:56:33 -0800 Subject: [Freeswitch-users] Executing Multiple Applications in parallel In-Reply-To: References: Message-ID: I created a spot where you can document this: http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_22:__Play_MOH_while_doing_a_database_lookup -MC On Sat, Jan 26, 2013 at 6:45 PM, JP wrote: > Thanks Anthony!! That works perfectly. This is what I did... > > > > > > > data="/tmp/WPJ_MUSIC_QUEUE_4.wav"/> > > > > This is pretty neat. It should be on the wiki. If someone can suggest the > best place to place this on the wiki, I can update it. > > Thank you, > JP > > On Fri, Jan 25, 2013 at 8:37 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> only non-ESL method to do what you want.... >> >> 1) make the dialplan use the FSAPI via variable expansion to call luarun >> on the script with the uuid as the first arg (this will go in a new thread) >> 2) play your long file >> 3) do what you need to do and then use the uuid to feed to uuid_transfer >> api >> >> >> >> >> On Fri, Jan 25, 2013 at 10:26 PM, Usama Zaidi wrote: >> >>> http://pastebin.freeswitch.org/20508 I just checked the link and it >>> works... >>> >>> >>> On Sat, Jan 26, 2013 at 5:44 AM, Usama Zaidi wrote: >>> >>>> Hey, >>>> >>>> I posted this earlier as a question, but I think this might be >>>> relevant. ttp://pastebin.freeswitch.org/20508. >>>> The workaround (hack) for my issue is to check the connection state >>>> ($con->connected()) every time you loop and reconnect and subscribe for >>>> events again when that returns false. >>>> >>>> On Sat, Jan 26, 2013 at 5:33 AM, < >>>> freeswitch-users-request at lists.freeswitch.org> wrote: >>>> >>>>> Send FreeSWITCH-users mailing list submissions to >>>>> freeswitch-users at lists.freeswitch.org >>>>> >>>>> To subscribe or unsubscribe via the World Wide Web, visit >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> or, via email, send a message with subject or body 'help' to >>>>> freeswitch-users-request at lists.freeswitch.org >>>>> >>>>> You can reach the person managing the list at >>>>> freeswitch-users-owner at lists.freeswitch.org >>>>> >>>>> When replying, please edit your Subject line so it is more specific >>>>> than "Re: Contents of FreeSWITCH-users digest..." >>>>> >>>>> Today's Topics: >>>>> >>>>> 1. Re: Executing multiple applications in parallel (Michael Collins) >>>>> 2. Re: Executing multiple applications in parallel (JP) >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: Michael Collins >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Fri, 25 Jan 2013 16:10:37 -0800 >>>>> Subject: Re: [Freeswitch-users] Executing multiple applications in >>>>> parallel >>>>> The problem you're running into is that you are trying to do two >>>>> different things inside of the thread that is controlling the call flow. >>>>> Playing sound files and running Lua scripts are synchronous in this >>>>> context. This seems more suited for an outbound event socket application. >>>>> Using the event socket you can control the call while doing other things. >>>>> >>>>> Just curious - does anyone have an ESL example of this kind of thing? >>>>> The wiki could definitely use this kind of example as it is quite useful. >>>>> I'll look around and see if I can find anything. >>>>> >>>>> -MC >>>>> >>>>> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >>>>> >>>>>> Avi, >>>>>> Thanks for the response. Unfortunately that did not work. This is >>>>>> what I did >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> The music only plays after the Lua script exits. I played around with >>>>>> the timer value, but that did not help either. Any other ideas? >>>>>> >>>>>> Thanks >>>>>> JP >>>>>> >>>>>> >>>>>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>>>>> >>>>>>> ESL gives you more control, especially to execute things in parallel. >>>>>>> If you want to do it within FS, that's a little more complicated, >>>>>>> since most things are sequential. >>>>>>> >>>>>>> However, if you use the async APIs, then I think it could work: >>>>>>> 1) sched_broadcast -- >>>>>>> schedule the playback of your file on the UUID of the channel, for +0, >>>>>>> meaning now. >>>>>>> 2) run your lua script >>>>>>> 3) Then stop (break) the playback on that uuid: break >>>>>>> 4) Continue with your execution. >>>>>>> >>>>>>> Please wikify your results and let us know how that works out! >>>>>>> -Avi >>>>>>> >>>>>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>>>>> >>>>>>>> I want to do the following... >>>>>>>> >>>>>>>> 1. Play a long music file from a dialplan. >>>>>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>>>>> some data dip. >>>>>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>>>>> music and continue in the dialplan >>>>>>>> >>>>>>>> How do I do this. Any sample code or pointers would be greatly >>>>>>>> appreciated. >>>>>>>> >>>>>>>> Thanks >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Michael S Collins >>>>> Twitter: @mercutioviz >>>>> http://www.FreeSWITCH.org >>>>> http://www.ClueCon.com >>>>> http://www.OSTAG.org >>>>> >>>>> >>>>> >>>>> ---------- Forwarded message ---------- >>>>> From: JP >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Date: Fri, 25 Jan 2013 16:32:50 -0800 >>>>> Subject: Re: [Freeswitch-users] Executing multiple applications in >>>>> parallel >>>>> MC, >>>>> Thanks for the info. Unfortunately outbound event socket application >>>>> is not an option for me right now. I was wondering if the following "ideas" >>>>> can be of any use to me( I know I am trying clutch at straws, but hope is >>>>> eternal :)) >>>>> 1. Use "bgapi". Although on the wiki I only see this used with >>>>> "originate". >>>>> 2. Can I use a conference to park the caller there while playing music >>>>> on another call leg of the conference. Meanwhile can I execute the Lua >>>>> script :)) >>>>> 3. Can I launch another Lua script from my original one asynchronously? >>>>> 4. Can I have a start up Lua script running in the background >>>>> listening for some custom events and I can have the main caller thread >>>>> generate the custom event and then start playing long music file. Once the >>>>> background Lua script is done with the data dip, it can just break the >>>>> music on the other channel. >>>>> >>>>> That's it from me. Someone help me from sinking... please. >>>>> >>>>> Thanks, >>>>> -JP >>>>> >>>>> >>>>> On Fri, Jan 25, 2013 at 4:10 PM, Michael Collins wrote: >>>>> >>>>>> The problem you're running into is that you are trying to do two >>>>>> different things inside of the thread that is controlling the call flow. >>>>>> Playing sound files and running Lua scripts are synchronous in this >>>>>> context. This seems more suited for an outbound event socket application. >>>>>> Using the event socket you can control the call while doing other things. >>>>>> >>>>>> Just curious - does anyone have an ESL example of this kind of thing? >>>>>> The wiki could definitely use this kind of example as it is quite useful. >>>>>> I'll look around and see if I can find anything. >>>>>> >>>>>> -MC >>>>>> >>>>>> >>>>>> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >>>>>> >>>>>>> Avi, >>>>>>> Thanks for the response. Unfortunately that did not work. This is >>>>>>> what I did >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> The music only plays after the Lua script exits. I played around >>>>>>> with the timer value, but that did not help either. Any other ideas? >>>>>>> >>>>>>> Thanks >>>>>>> JP >>>>>>> >>>>>>> >>>>>>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>>>>>> >>>>>>>> ESL gives you more control, especially to execute things >>>>>>>> in parallel. >>>>>>>> If you want to do it within FS, that's a little more complicated, >>>>>>>> since most things are sequential. >>>>>>>> >>>>>>>> However, if you use the async APIs, then I think it could work: >>>>>>>> 1) sched_broadcast -- >>>>>>>> schedule the playback of your file on the UUID of the channel, for +0, >>>>>>>> meaning now. >>>>>>>> 2) run your lua script >>>>>>>> 3) Then stop (break) the playback on that uuid: break >>>>>>>> 4) Continue with your execution. >>>>>>>> >>>>>>>> Please wikify your results and let us know how that works out! >>>>>>>> -Avi >>>>>>>> >>>>>>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>>>>>> >>>>>>>>> I want to do the following... >>>>>>>>> >>>>>>>>> 1. Play a long music file from a dialplan. >>>>>>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>>>>>> some data dip. >>>>>>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>>>>>> music and continue in the dialplan >>>>>>>>> >>>>>>>>> How do I do this. Any sample code or pointers would be greatly >>>>>>>>> appreciated. >>>>>>>>> >>>>>>>>> Thanks >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Michael S Collins >>>>>> Twitter: @mercutioviz >>>>>> http://www.FreeSWITCH.org >>>>>> http://www.ClueCon.com >>>>>> http://www.OSTAG.org >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> I'd love to change the world, but they wont gimme the source code to it >>>> >>> >>> >>> >>> -- >>> I'd love to change the world, but they wont gimme the source code to it >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/3c2ca50b/attachment-0001.html From schoch+freeswitch.org at xwin32.com Mon Jan 28 21:24:53 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 28 Jan 2013 10:24:53 -0800 Subject: [Freeswitch-users] Book recommendation: Exploding the Phone Message-ID: This is off-topic, but many of you would like the book "Exploding the Phone" by my old college buddy, Phil Lapsley. It's about the history of hacking the phone system. It's available on Amazon.com, and he's doing a tour in February. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/51154631/attachment.html From msc at freeswitch.org Mon Jan 28 21:43:12 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Jan 2013 10:43:12 -0800 Subject: [Freeswitch-users] Patton FXO configuration on Freeswitch - more details In-Reply-To: <51062C9C.8020701@asistar.it> References: <5102AAF6.2070201@asistar.it> <51062C9C.8020701@asistar.it> Message-ID: > Hi Michael, > thanks for the reply. > > So if I register in directory as a phone how do I get out of the > gateway, for example through a bridge? > > Alessio > User directory is for inbound registrations and SIP auth challenges for devices connecting to the CudaTel. Gateways are the opposite - they are for auth challenges and registrations going from the CudaTel to other devices. To dial out a gateway you just use the syntax: -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/4adf47a7/attachment.html From msc at freeswitch.org Mon Jan 28 21:44:33 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 28 Jan 2013 10:44:33 -0800 Subject: [Freeswitch-users] Music on Hold In-Reply-To: References: <1359364817390-7586777.post@n2.nabble.com> Message-ID: You need to set the variable in the dialplan based upon whether someone in the group is making or receiving the call. How have you defined your group? Is it in conf/directory/default.xml? -MC On Mon, Jan 28, 2013 at 1:37 AM, madan.mallikarjun < madan.mallikarjun at outlook.com> wrote: > hi > > this actually changes in the dial plan but i need to change it only for a > group. > > thanks > > ------------------------------ > Date: Mon, 28 Jan 2013 01:34:55 -0800 > From: [hidden email] > To: [hidden email] > Subject: Re: Music on Hold > > See http://wiki.freeswitch.org/wiki/Variable_hold_music > > > On 1/28/13 3:20 AM, "madan.mallikarjun" <[hidden email]> > > wrote: > > > HI all, > > > > how do we change the music on hold for a particular group ie assuming i > have > > 2 groups for which i do not want to play the default MOH for 1 of the > groups > > > > > > thanks > > > > > > > > -- > > View this message in context: > > > http://freeswitch-users.2379917.n2.nabble.com/Music-on-Hold-tp7586777.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > [hidden email] > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > [hidden email] > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > [hidden email] > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > > http://freeswitch-users.2379917.n2.nabble.com/Music-on-Hold-tp7586777p7586778.html > To unsubscribe from Music on Hold, click here. > NAML > > ------------------------------ > View this message in context: RE: Music on Hold > > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/a3dbc465/attachment.html From anthony.minessale at gmail.com Mon Jan 28 22:12:46 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Jan 2013 13:12:46 -0600 Subject: [Freeswitch-users] Webrtc In-Reply-To: <1359381458946-7586790.post@n2.nabble.com> References: <1359381458946-7586790.post@n2.nabble.com> Message-ID: I would be careful with sipml5. All of the code related to it is GPLv3 which is the new version of the GPL that goes viral over a socket. So if you make a product from this code and sell it as a service, you will be bound to the GPL license and forced to surrender your platform. See the Web Services section in http://neptune.billgatliff.com/GPL3.html Also the project has changed its goals towards making a converter to change webrtc based connections into regular SIP for people who do not wish to implement the webrtc profile....... On Mon, Jan 28, 2013 at 7:57 AM, soni4m wrote: > Hi Anthony, > Can you please explain how the overall solution will look like once this > webrtc support in FS effort completes ? > Will it be the way asterisk is used in many of these demos (One example > here > < > http://highsecurity.blogspot.in/2012/12/webrtc-and-asterisk-11-using-sipml5.html > > > ) In a sense that sipml5 (SIP stack in the browser) is necessarily there > on > the browser side. My interest here is to understand the topology here how > FS will help connecting the legacy SIP devices with browser and what will > come into the picture in between. > Thanks > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Webrtc-tp7580875p7586790.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/84c1c028/attachment-0001.html From shaheryarkh at gmail.com Mon Jan 28 23:16:53 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Mon, 28 Jan 2013 21:16:53 +0100 Subject: [Freeswitch-users] How to auto save voicemail message? In-Reply-To: References: Message-ID: Thanks, i got it working now. The user greeting was set to beep with 10 second silence afterwards, which confused me with voicemail beep, an interesting mistake from my colleague. Thank you. On Mon, Jan 28, 2013 at 12:22 PM, Avi Marcus wrote: > Get a log in FS of the call. Maybe it's a permission issue. The logs > should help. > > -Avi > (This message was painstakingly thumbed out on my mobile, so apologies for > brevity and errors.) > > On Jan 28, 2013 12:10 PM, "Muhammad Shahzad" > wrote: > > For some reason its not working for me. I have FS behind OpenSIPS, serving > only voicemail, so when call comes in on FS, i simply send it to voicemail > directly. Here is my dial plan. > > > > > > > > > > data="transfer_fallback_extension=my_dummy_operator"/> > > > > > > > > > > > But its not working accordingly. Do you suspect anything wrong in it? > Which might be causing it? > > Thank you. > > > > > On Mon, Jan 28, 2013 at 8:30 AM, Avi Marcus wrote: > > > > If you call FS's voice... > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/7f77cff4/attachment.html From jaykris at gmail.com Mon Jan 28 23:27:33 2013 From: jaykris at gmail.com (JP) Date: Mon, 28 Jan 2013 12:27:33 -0800 Subject: [Freeswitch-users] Executing Multiple Applications in parallel In-Reply-To: References: Message-ID: Done! Thanks to all for taking the time to respond to the issue I was facing. JP On Mon, Jan 28, 2013 at 9:56 AM, Michael Collins wrote: > I created a spot where you can document this: > > > http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_22:__Play_MOH_while_doing_a_database_lookup > > -MC > > > On Sat, Jan 26, 2013 at 6:45 PM, JP wrote: > >> Thanks Anthony!! That works perfectly. This is what I did... >> >> >> >> >> >> >> > data="/tmp/WPJ_MUSIC_QUEUE_4.wav"/> >> >> >> >> This is pretty neat. It should be on the wiki. If someone can suggest the >> best place to place this on the wiki, I can update it. >> >> Thank you, >> JP >> >> On Fri, Jan 25, 2013 at 8:37 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> only non-ESL method to do what you want.... >>> >>> 1) make the dialplan use the FSAPI via variable expansion to call luarun >>> on the script with the uuid as the first arg (this will go in a new thread) >>> 2) play your long file >>> 3) do what you need to do and then use the uuid to feed to uuid_transfer >>> api >>> >>> >>> >>> >>> On Fri, Jan 25, 2013 at 10:26 PM, Usama Zaidi wrote: >>> >>>> http://pastebin.freeswitch.org/20508 I just checked the link and it >>>> works... >>>> >>>> >>>> On Sat, Jan 26, 2013 at 5:44 AM, Usama Zaidi wrote: >>>> >>>>> Hey, >>>>> >>>>> I posted this earlier as a question, but I think this might be >>>>> relevant. ttp://pastebin.freeswitch.org/20508. >>>>> The workaround (hack) for my issue is to check the connection state >>>>> ($con->connected()) every time you loop and reconnect and subscribe for >>>>> events again when that returns false. >>>>> >>>>> On Sat, Jan 26, 2013 at 5:33 AM, < >>>>> freeswitch-users-request at lists.freeswitch.org> wrote: >>>>> >>>>>> Send FreeSWITCH-users mailing list submissions to >>>>>> freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> To subscribe or unsubscribe via the World Wide Web, visit >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> or, via email, send a message with subject or body 'help' to >>>>>> freeswitch-users-request at lists.freeswitch.org >>>>>> >>>>>> You can reach the person managing the list at >>>>>> freeswitch-users-owner at lists.freeswitch.org >>>>>> >>>>>> When replying, please edit your Subject line so it is more specific >>>>>> than "Re: Contents of FreeSWITCH-users digest..." >>>>>> >>>>>> Today's Topics: >>>>>> >>>>>> 1. Re: Executing multiple applications in parallel (Michael >>>>>> Collins) >>>>>> 2. Re: Executing multiple applications in parallel (JP) >>>>>> >>>>>> >>>>>> ---------- Forwarded message ---------- >>>>>> From: Michael Collins >>>>>> To: FreeSWITCH Users Help >>>>>> Cc: >>>>>> Date: Fri, 25 Jan 2013 16:10:37 -0800 >>>>>> Subject: Re: [Freeswitch-users] Executing multiple applications in >>>>>> parallel >>>>>> The problem you're running into is that you are trying to do two >>>>>> different things inside of the thread that is controlling the call flow. >>>>>> Playing sound files and running Lua scripts are synchronous in this >>>>>> context. This seems more suited for an outbound event socket application. >>>>>> Using the event socket you can control the call while doing other things. >>>>>> >>>>>> Just curious - does anyone have an ESL example of this kind of thing? >>>>>> The wiki could definitely use this kind of example as it is quite useful. >>>>>> I'll look around and see if I can find anything. >>>>>> >>>>>> -MC >>>>>> >>>>>> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >>>>>> >>>>>>> Avi, >>>>>>> Thanks for the response. Unfortunately that did not work. This is >>>>>>> what I did >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> The music only plays after the Lua script exits. I played around >>>>>>> with the timer value, but that did not help either. Any other ideas? >>>>>>> >>>>>>> Thanks >>>>>>> JP >>>>>>> >>>>>>> >>>>>>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>>>>>> >>>>>>>> ESL gives you more control, especially to execute things >>>>>>>> in parallel. >>>>>>>> If you want to do it within FS, that's a little more complicated, >>>>>>>> since most things are sequential. >>>>>>>> >>>>>>>> However, if you use the async APIs, then I think it could work: >>>>>>>> 1) sched_broadcast -- >>>>>>>> schedule the playback of your file on the UUID of the channel, for +0, >>>>>>>> meaning now. >>>>>>>> 2) run your lua script >>>>>>>> 3) Then stop (break) the playback on that uuid: break >>>>>>>> 4) Continue with your execution. >>>>>>>> >>>>>>>> Please wikify your results and let us know how that works out! >>>>>>>> -Avi >>>>>>>> >>>>>>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>>>>>> >>>>>>>>> I want to do the following... >>>>>>>>> >>>>>>>>> 1. Play a long music file from a dialplan. >>>>>>>>> 2. While the music is playing, I want to launch a Lua script to do >>>>>>>>> some data dip. >>>>>>>>> 3. Once the data is fetched and Lua exits, I want to interrupt the >>>>>>>>> music and continue in the dialplan >>>>>>>>> >>>>>>>>> How do I do this. Any sample code or pointers would be greatly >>>>>>>>> appreciated. >>>>>>>>> >>>>>>>>> Thanks >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Michael S Collins >>>>>> Twitter: @mercutioviz >>>>>> http://www.FreeSWITCH.org >>>>>> http://www.ClueCon.com >>>>>> http://www.OSTAG.org >>>>>> >>>>>> >>>>>> >>>>>> ---------- Forwarded message ---------- >>>>>> From: JP >>>>>> To: FreeSWITCH Users Help >>>>>> Cc: >>>>>> Date: Fri, 25 Jan 2013 16:32:50 -0800 >>>>>> Subject: Re: [Freeswitch-users] Executing multiple applications in >>>>>> parallel >>>>>> MC, >>>>>> Thanks for the info. Unfortunately outbound event socket application >>>>>> is not an option for me right now. I was wondering if the following "ideas" >>>>>> can be of any use to me( I know I am trying clutch at straws, but hope is >>>>>> eternal :)) >>>>>> 1. Use "bgapi". Although on the wiki I only see this used with >>>>>> "originate". >>>>>> 2. Can I use a conference to park the caller there while playing >>>>>> music on another call leg of the conference. Meanwhile can I execute the >>>>>> Lua script :)) >>>>>> 3. Can I launch another Lua script from my original one >>>>>> asynchronously? >>>>>> 4. Can I have a start up Lua script running in the background >>>>>> listening for some custom events and I can have the main caller thread >>>>>> generate the custom event and then start playing long music file. Once the >>>>>> background Lua script is done with the data dip, it can just break the >>>>>> music on the other channel. >>>>>> >>>>>> That's it from me. Someone help me from sinking... please. >>>>>> >>>>>> Thanks, >>>>>> -JP >>>>>> >>>>>> >>>>>> On Fri, Jan 25, 2013 at 4:10 PM, Michael Collins wrote: >>>>>> >>>>>>> The problem you're running into is that you are trying to do two >>>>>>> different things inside of the thread that is controlling the call flow. >>>>>>> Playing sound files and running Lua scripts are synchronous in this >>>>>>> context. This seems more suited for an outbound event socket application. >>>>>>> Using the event socket you can control the call while doing other things. >>>>>>> >>>>>>> Just curious - does anyone have an ESL example of this kind of >>>>>>> thing? The wiki could definitely use this kind of example as it is quite >>>>>>> useful. I'll look around and see if I can find anything. >>>>>>> >>>>>>> -MC >>>>>>> >>>>>>> >>>>>>> On Fri, Jan 25, 2013 at 3:00 PM, JP wrote: >>>>>>> >>>>>>>> Avi, >>>>>>>> Thanks for the response. Unfortunately that did not work. This is >>>>>>>> what I did >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> The music only plays after the Lua script exits. I played around >>>>>>>> with the timer value, but that did not help either. Any other ideas? >>>>>>>> >>>>>>>> Thanks >>>>>>>> JP >>>>>>>> >>>>>>>> >>>>>>>> On Fri, Jan 25, 2013 at 1:22 AM, Avi Marcus wrote: >>>>>>>> >>>>>>>>> ESL gives you more control, especially to execute things >>>>>>>>> in parallel. >>>>>>>>> If you want to do it within FS, that's a little more complicated, >>>>>>>>> since most things are sequential. >>>>>>>>> >>>>>>>>> However, if you use the async APIs, then I think it could work: >>>>>>>>> 1) sched_broadcast -- >>>>>>>>> schedule the playback of your file on the UUID of the channel, for +0, >>>>>>>>> meaning now. >>>>>>>>> 2) run your lua script >>>>>>>>> 3) Then stop (break) the playback on that uuid: break >>>>>>>>> 4) Continue with your execution. >>>>>>>>> >>>>>>>>> Please wikify your results and let us know how that works out! >>>>>>>>> -Avi >>>>>>>>> >>>>>>>>> On Fri, Jan 25, 2013 at 3:34 AM, JP wrote: >>>>>>>>> >>>>>>>>>> I want to do the following... >>>>>>>>>> >>>>>>>>>> 1. Play a long music file from a dialplan. >>>>>>>>>> 2. While the music is playing, I want to launch a Lua script to >>>>>>>>>> do some data dip. >>>>>>>>>> 3. Once the data is fetched and Lua exits, I want to interrupt >>>>>>>>>> the music and continue in the dialplan >>>>>>>>>> >>>>>>>>>> How do I do this. Any sample code or pointers would be greatly >>>>>>>>>> appreciated. >>>>>>>>>> >>>>>>>>>> Thanks >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Michael S Collins >>>>>>> Twitter: @mercutioviz >>>>>>> http://www.FreeSWITCH.org >>>>>>> http://www.ClueCon.com >>>>>>> http://www.OSTAG.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> I'd love to change the world, but they wont gimme the source code to >>>>> it >>>>> >>>> >>>> >>>> >>>> -- >>>> I'd love to change the world, but they wont gimme the source code to it >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/bad8efb3/attachment-0001.html From marketing at cluecon.com Mon Jan 28 23:29:51 2013 From: marketing at cluecon.com (Michael Collins) Date: Mon, 28 Jan 2013 12:29:51 -0800 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Hello all! I wanted to start off by giving you all some good news: Our friends at Newegg have successfully slain a Patent Troll named Soverain. This NPE (non-producing entity) acquired some questionable patents and started suing just about anybody with an online shopping cart on their website. To make a long story short: Newegg fought them in court instead of settling and in the process got all three patents invalidated! More details available at the Ars Technica article. In any event, those of us who do software for a living, especially in the US, are well aware of the dangerous patent situation we have. Hattip to Newegg for doing us all a favor. More good news! We have released a few more ClueCon 2012 videos: * Open Source is revamping telecom. You want in. Learn from our wins and misses.- Darren Schreiber * HOMER in the cloud. - Alexandr Dubovikov * OSTAG - Who we are, what we do, and why.- Raymond Chandler I will be uploading the slides for the talks shortly. In other ClueCon news we are gearing up for the 2013 event. We look forward to hearing from prospective speakers, sponsors, and attendees. As a reminder, we are at the same hotel this year and the event runs from August 6th through the 8th. Stay tuned for more information. We have an annual tradition that is coming up next week: buy the developers dinner! Quite simply, we invite the community to throw some money into the hat and then the developers use it to buy dinner. The easiest way to donate is to click the Donate button on the main FreeSWITCH website. (Right-hand side, under the ClueCon 2013 logo.) Do a Paypal transaction, mark it person (a gift is fine) and in the message to the recipient mention that the money is for the developer dinner. We appreciate the community doing this again for us this year. Thank you! We'll talk to you this Wednesday on our weekly conference call. Last week was an open discussion. This week I would will be doing an event socket and ESL review. More people are using these great tools and we are seeing some of the same questions being asked so we want to make sure that we get this information out there for everyone to use. Thanks again for being a great community and we will talk to you again next week. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/ffa6d152/attachment.html From jpyle at fidelityvoice.com Tue Jan 29 01:41:10 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Mon, 28 Jan 2013 17:41:10 -0500 Subject: [Freeswitch-users] unexplained RAM usage increase Message-ID: Hello, I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode LX processor (i386). I can sustain 30 concurrent calls averaging around 70% CPU utilization by the freeswitch process, measured by top. Bypass media and proxy media are disabled. PCMU is forced on both endpoints (no transcoding). The problem is the RAM usage over time. The board has 256M. Idle, freeswitch occupies around 4% after a fresh restart. A minute or so after 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, 13.6%. After 60 minutes, near 65%. Disconnecting the calls returns the RAM usage to 6-8%. I've not tried to troubleshoot an issue like this before. Is valgrind the next step, or would something else make more sense? Regards, Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/737ed31b/attachment.html From yungwei at resolvity.com Tue Jan 29 01:43:30 2013 From: yungwei at resolvity.com (Yungwei Chen) Date: Mon, 28 Jan 2013 17:43:30 -0500 Subject: [Freeswitch-users] voices in the recordings are out of sync In-Reply-To: <33095823FD21DF429B481B5163264B799F879B1B1A@VMBX102.ihostexchange.net> References: <33095823FD21DF429B481B5163264B799F6D8752B6@VMBX102.ihostexchange.net> <33095823FD21DF429B481B5163264B799F879B1B1A@VMBX102.ihostexchange.net> Message-ID: <33095823FD21DF429B481B5163264B799F879B1C8F@VMBX102.ihostexchange.net> Tested against the following version today, and the problem is still there. FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git a424765 2013-01-04 15:45:59Z) From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen Sent: Friday, January 25, 2013 5:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] voices in the recordings are out of sync Hi, I did a quick test against freeswitch-1.2.5.3 on CentOS 5.8, but the problem is still there. In the recording, the caller speaks way faster than he did. Here's the version of freeswitch: FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z) Here's the set of steps: 1. In dialplan/default/resolvity.xml, add the following extension: 2. Create scripts/test.js with the following content. This js file will read numbers sequentially starting from 0. session.execute("record_session", "/tmp/test.wav"); for (var i=0; i<100; i++) { session.speak("flite", "kal", i+''); } session.hangup(16); 3. reloadxml 4. dial 0009 from a registered SIP client, and then repeat each number you heard. 5. Listen to the recording, /tmp/test.wav. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, December 18, 2012 8:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] voices in the recordings are out of sync Latest version of FreeSWITCH has some updates that may fix this issue. I would update to 1.2.5.3 ASAP. -MC On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen > wrote: Hi, I found one issue that voices are always out of sync in the recordings. I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed from yum. I am having trouble installing the latest version from source due to an error: Autoconf version 2.62 or higher is required. It would be nice if someone can reproduce this issue against HEAD. Thanks. Here're the steps to reproduce it. The idea is to call a phone number and then bridge to another phone number while the entire session is being recorded. 1. In dialplan/public.xml, make sure you have a dialplan to handle any 10 digit phone numbers. 2. In dialplan/default/main.xml, make sure you have an extension to handle the call in the default context. 3. In sip_profiles/external/gateways.xml, make sure you have a gateway that allows you to make an outbound call. 4. make a call to one of the allowed 10-digit phone numbers in your environment. 5. Once the call is answered, the caller shall start to count from 1 to 60 with some pause after each number. 6. The callee shall repeat each number he/she heard from the caller. 7. You should be able to hear that 2 voices in the recoridng (/tmp/rec.wav) are out of sync. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/bae25b9f/attachment-0001.html From Sirish.MasurMohan at oa.com.au Tue Jan 29 01:45:53 2013 From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan) Date: Tue, 29 Jan 2013 09:45:53 +1100 Subject: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error In-Reply-To: References: <965759A53E43FE439E43565A7715E5F0591C085DB2@oa-exchange1.oa.com.au> Message-ID: <965759A53E43FE439E43565A7715E5F0591C085DCF@oa-exchange1.oa.com.au> Hi Michael, I have put the logs and SIP trace in the pastebin: http://pastebin.freeswitch.org/20514 I haven't created extensions for the SNOM phones in freeswitch, as the ACL for registration and incoming are Open - could this be a reason for the error? acl.conf.xml - sip_profiles/internal.xml With regards, Sirish From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, 26 January 2013 7:09 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error I confess that I haven't seen this specific error before. Could you get the SIP trace of this call and put it on pastebin as well? That might give you the information you need to crack the case. -MC On Thu, Jan 24, 2013 at 10:41 PM, Sirish Masur Mohan > wrote: Hello All, I think I am missing something obvious here, but I am having trouble trying to make calls between two SNOM phones. I have an open ACL for registration and I keep getting BEARERCAPABILITY_NOTAUTH error when I try to make calls. Would appreciate any inputs on how to debug this! Logs are at : http://pastebin.freeswitch.org/20504 With regards, Sirish _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/75d2f148/attachment.html From lconroy at insensate.co.uk Tue Jan 29 01:51:50 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Mon, 28 Jan 2013 22:51:50 +0000 Subject: [Freeswitch-users] Webrtc license In-Reply-To: References: <1359381458946-7586790.post@n2.nabble.com> Message-ID: Hi Anthony, folks, personally, I agree that GPL is a pile of pox, but the ebola version is called Affero. I did look at the sipml5 home, and the web site appears to claim that it is covered by GPL3. Note -- not GNU Affero. Did a grep and can't find Affero anywhere in their source. I had thought that it was the combination with *any* GNU Affero licensed lump that makes it viral over the socket. [Section 13 of the GNUv3 license spells that out]. Thus, if none of it is Affero-licensed, it's just evil and wrong, not level 5 toxic. Afferophobia had me looking at their web site in double quick time. panic over. all the best, Lawrence p.s. agreed -- the potentially useful bit is the webrtc2sip converter gubbins On 28 Jan 2013, at 19:12, Anthony Minessale wrote: > I would be careful with sipml5. All of the code related to it is GPLv3 > which is the new version of the GPL that goes viral over a socket. So if > you make a product from this code and sell it as a service, you will be > bound to the GPL license and forced to surrender your platform. > > See the Web Services section in > http://neptune.billgatliff.com/GPL3.html > > Also the project has changed its goals towards making a converter to change > webrtc based connections into regular SIP for people who do not wish to > implement the webrtc profile....... > > > > On Mon, Jan 28, 2013 at 7:57 AM, soni4m wrote: > >> Hi Anthony, >> Can you please explain how the overall solution will look like once this >> webrtc support in FS effort completes ? >> Will it be the way asterisk is used in many of these demos (One example >> here >> >> ) In a sense that sipml5 (SIP stack in the browser) is necessarily there >> on >> the browser side. My interest here is to understand the topology here how >> FS will help connecting the legacy SIP devices with browser and what will >> come into the picture in between. >> Thanks >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Webrtc-tp7580875p7586790.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. > -- > Anthony Minessale II From krice at freeswitch.org Tue Jan 29 01:58:09 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 28 Jan 2013 16:58:09 -0600 Subject: [Freeswitch-users] voices in the recordings are out of sync In-Reply-To: <33095823FD21DF429B481B5163264B799F879B1C8F@VMBX102.ihostexchange.net> Message-ID: Ok... You should be testing against MASTER... If you are not testing again Master It is not going to get fixed... All patches go there first... If you can duplicate there, then open a ticket. This is the only way these sort of things get fixed On 1/28/13 4:43 PM, "Yungwei Chen" wrote: > Tested against the following version today, and the problem is still there. > FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git a424765 > 2013-01-04 15:45:59Z) > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei > Chen > Sent: Friday, January 25, 2013 5:07 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] voices in the recordings are out of sync > > Hi, > > I did a quick test against freeswitch-1.2.5.3 on CentOS 5.8, but the problem > is still there. > In the recording, the caller speaks way faster than he did. > > Here's the version of freeswitch: > FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 > 2012-12-29 00:17:59Z) > > Here's the set of steps: > 1. In dialplan/default/resolvity.xml, add the following extension: > > > > > > > > > 2. Create scripts/test.js with the following content. This js file will read > numbers sequentially starting from 0. > session.execute("record_session", "/tmp/test.wav"); > for (var i=0; i<100; i++) > { > session.speak("flite", "kal", i+''); > } > session.hangup(16); > > 3. reloadxml > 4. dial 0009 from a registered SIP client, and then repeat each number you > heard. > 5. Listen to the recording, /tmp/test.wav. > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael > Collins > Sent: Tuesday, December 18, 2012 8:26 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] voices in the recordings are out of sync > > Latest version of FreeSWITCH has some updates that may fix this issue. I would > update to 1.2.5.3 ASAP. > -MC > > On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen wrote: > Hi, > > I found one issue that voices are always out of sync in the recordings. > I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed from > yum. > I am having trouble installing the latest version from source due to an error: > Autoconf version 2.62 or higher is required. > It would be nice if someone can reproduce this issue against HEAD. Thanks. > > Here're the steps to reproduce it. The idea is to call a phone number and then > bridge to another phone number while the entire session is being recorded. > 1. In dialplan/public.xml, make sure you have a dialplan to handle any 10 > digit phone numbers. > > > > > > > 2. In dialplan/default/main.xml, make sure you have an extension to handle the > call in the default context. > > > > > data="/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_nu > mber}.wav"/> > data="{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890"/> > > > > > 3. In sip_profiles/external/gateways.xml, make sure you have a gateway that > allows you to make an outbound call. > > > > > > > > > > > > > > 4. make a call to one of the allowed 10-digit phone numbers in your > environment. > 5. Once the call is answered, the caller shall start to count from 1 to 60 > with some pause after each number. > 6. The callee shall repeat each number he/she heard from the caller. > 7. You should be able to hear that 2 voices in the recoridng (/tmp/rec.wav) > are out of sync. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/6bc327a0/attachment-0001.html From grcamauer at gmail.com Tue Jan 29 02:30:36 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Mon, 28 Jan 2013 20:30:36 -0300 Subject: [Freeswitch-users] fs_encode and segmentation fault Message-ID: I am trying the following command: fs_encode -l mod_sangoma_codec greeting.wav greeting.G729 which returns: 31663 Segmentation fault If I try to convert to PCMU it works fine: fs_encode -l mod_sangoma_codec greeting.wav greeting.PCMU Making and receiving calls with G729 works fine. A similar command on a machine with a DIGIUM card works fine: fs_encode -l mod_dahdi_codec greeting.wav greeting.G729 The only difference I see is the card and the module being loaded. Any idea of what could be happenning? Do I need to load some other module for the Sangoma card? Thanks in advance -- Guillermo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/c56bda73/attachment.html From kris at kriskinc.com Tue Jan 29 02:45:10 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 28 Jan 2013 18:45:10 -0500 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: Out of curiosity does your distro use uclibc, eglibc, or glibc? On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle wrote: > Hello, > > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode LX > processor (i386). I can sustain 30 concurrent calls averaging around 70% > CPU utilization by the freeswitch process, measured by top. Bypass media > and proxy media are disabled. PCMU is forced on both endpoints (no > transcoding). > > The problem is the RAM usage over time. The board has 256M. Idle, > freeswitch occupies around 4% after a fresh restart. A minute or so after > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, 13.6%. > After 60 minutes, near 65%. Disconnecting the calls returns the RAM usage > to 6-8%. > > I've not tried to troubleshoot an issue like this before. Is valgrind the > next step, or would something else make more sense? > > > Regards, > Jeff > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From steveayre at gmail.com Tue Jan 29 03:57:06 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 29 Jan 2013 00:57:06 +0000 Subject: [Freeswitch-users] fs_encode and segmentation fault In-Reply-To: References: Message-ID: Can you reproduce this on Git master head? Please enable coredumps (ulimit -c unlimited) and collect a backtrace, then open a Jira bug ticket and attach the backtrace. http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 http://jira.freeswitch.org/ -Steve On 28 January 2013 23:30, Guillermo Ruiz Camauer wrote: > > > I am trying the following command: > fs_encode -l mod_sangoma_codec greeting.wav greeting.G729 > > which returns: > > 31663 Segmentation fault > > If I try to convert to PCMU it works fine: > fs_encode -l mod_sangoma_codec greeting.wav greeting.PCMU > > Making and receiving calls with G729 works fine. > > A similar command on a machine with a DIGIUM card works fine: > fs_encode -l mod_dahdi_codec greeting.wav greeting.G729 > > The only difference I see is the card and the module being loaded. Any > idea of what could be happenning? Do I need to load some other module for > the Sangoma card? > > Thanks in advance > > -- > Guillermo > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/b015c999/attachment.html From jpyle at fidelityvoice.com Tue Jan 29 04:43:03 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Mon, 28 Jan 2013 20:43:03 -0500 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. - Jeff On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner wrote: > Out of curiosity does your distro use uclibc, eglibc, or glibc? > > On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle > wrote: > > Hello, > > > > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode > LX > > processor (i386). I can sustain 30 concurrent calls averaging around 70% > > CPU utilization by the freeswitch process, measured by top. Bypass media > > and proxy media are disabled. PCMU is forced on both endpoints (no > > transcoding). > > > > The problem is the RAM usage over time. The board has 256M. Idle, > > freeswitch occupies around 4% after a fresh restart. A minute or so > after > > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, > 13.6%. > > After 60 minutes, near 65%. Disconnecting the calls returns the RAM > usage > > to 6-8%. > > > > I've not tried to troubleshoot an issue like this before. Is valgrind > the > > next step, or would something else make more sense? > > > > > > Regards, > > Jeff > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/5017fc8e/attachment.html From ssoni at lifesize.com Tue Jan 29 05:16:09 2013 From: ssoni at lifesize.com (Sanjay Soni) Date: Mon, 28 Jan 2013 20:16:09 -0600 Subject: [Freeswitch-users] Webrtc Message-ID: Thanks. I also had this license thing in mind and didnt like this sipml business! May be i should have framed my question differently. So, what i want to know is how the webrtc solution will look like with fs ? Because frankly there is no information about it anywhere except some of the mailing list old emails which also doesnt give this information. Anthony Minessale wrote: I would be careful with sipml5. All of the code related to it is GPLv3 which is the new version of the GPL that goes viral over a socket. So if you make a product from this code and sell it as a service, you will be bound to the GPL license and forced to surrender your platform. See the Web Services section in http://neptune.billgatliff.com/GPL3.html Also the project has changed its goals towards making a converter to change webrtc based connections into regular SIP for people who do not wish to implement the webrtc profile....... On Mon, Jan 28, 2013 at 7:57 AM, soni4m > wrote: Hi Anthony, Can you please explain how the overall solution will look like once this webrtc support in FS effort completes ? Will it be the way asterisk is used in many of these demos (One example here ) In a sense that sipml5 (SIP stack in the browser) is necessarily there on the browser side. My interest here is to understand the topology here how FS will help connecting the legacy SIP devices with browser and what will come into the picture in between. Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Webrtc-tp7580875p7586790.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/008d66a9/attachment-0001.html From jaganthoutam at gmail.com Tue Jan 29 06:07:44 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Tue, 29 Jan 2013 08:37:44 +0530 Subject: [Freeswitch-users] Need Help On mod_xml_curl In-Reply-To: References: Message-ID: hi john, #!/usr/bin/perl #!/usr/bin/perl use warnings; use strict; use XML::Writer; use XML::Writer::String; my $s = XML::Writer::String->new(); my $writer = new XML::Writer( OUTPUT => $s,DATA_MODE => 1, DATA_INDENT => 4); $writer->xmlDecl('UTF-8', 'no'); $writer->startTag('document', type=>'freeswitch/xml'); $writer->startTag('section',name=>'dialplan',description=>'RE Dial Plan For FreeSwitch'); $writer->startTag('context',name=>'public'); $writer->startTag('condition',field=>'destination_number',expression=>'^(80\d{2})$'); $writer->startTag('action',application=>'set',data=>'dialed_extension=$1'); $writer->startTag('action',application=>'bridge',data=>'sofia/gateway/peer_asterisk/$1'); $writer->endTag('action'); $writer->endTag('action'); $writer->endTag('condition'); $writer->endTag('context'); $writer->endTag('section'); $writer->endTag('document'); $writer->end(); print $s->value(); i Think This can help you Thanks Jagadish On 28 January 2013 22:40, Cal Leeming [Simplicity Media Ltd] wrote: > Hi Jonathan, > > On a separate side note, that code is particularly unclean. > > You may want to consider either building the XML response using an XML > library, or use a template parsing language so you are not placing raw XML > in your source. > > In regards to your original email, the link Steven has provided is a good > starting point. > > Hope this helps. > > Cal > > On Sun, Jan 27, 2013 at 1:16 PM, johnthan123 wrote: >> >> HI All, >> >> >> Can You Please any one help me To Write Directory and Dialplan and In >> XML_CURL Like Below >> >> #################CODE######################## >> #!/usr/bin/perl >> use strict; >> use warnings; >> use Data::Dump qw(dump); >> use CGI; >> my $q = CGI->new; >> my $section = $q->param('section'); >> >> if ($section eq 'directory'){ >> >> $XML_STRING2 = ' >> >> >>
>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>
>> '; >> } >> elsif ( $section eq 'dialplan' ) { >> $XML_STRING = ' >>
>> >> >> >> >> >> > data="sofia/gateway/peer_asterisk/$1"/> >> >> >> >>
>> >> '; >> } >> >> ###########################END############################# >> >> This is Working for Me But i cant able to take the Database values into >> XML file >> >> >> Thanks In Advance >> >> >> Thanks >> >> J :: T >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Tue Jan 29 06:11:05 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Jan 2013 21:11:05 -0600 Subject: [Freeswitch-users] Webrtc In-Reply-To: References: Message-ID: We are still in the process of defining that. The first goal will be to support the media scheme required by the standards. There are many RFC earmarked into this that are requirements. The signaling is open so we'll start with SIP to get things moving then entertain other methods after that. On Mon, Jan 28, 2013 at 8:16 PM, Sanjay Soni wrote: > Thanks. I also had this license thing in mind and didnt like this sipml business! > May be i should have framed my question differently. So, what i want to know is how the webrtc solution will look like with fs ? Because frankly there is no information about it anywhere except some of the mailing list old emails which also doesnt give this information. > > Anthony Minessale wrote: > > > I would be careful with sipml5. All of the code related to it is GPLv3 > which is the new version of the GPL that goes viral over a socket. So if > you make a product from this code and sell it as a service, you will be > bound to the GPL license and forced to surrender your platform. > > See the Web Services section in > http://neptune.billgatliff.com/GPL3.html > > Also the project has changed its goals towards making a converter to > change webrtc based connections into regular SIP for people who do not wish > to implement the webrtc profile....... > > > > On Mon, Jan 28, 2013 at 7:57 AM, soni4m wrote: > >> Hi Anthony, >> Can you please explain how the overall solution will look like once this >> webrtc support in FS effort completes ? >> Will it be the way asterisk is used in many of these demos (One example >> here >> < >> http://highsecurity.blogspot.in/2012/12/webrtc-and-asterisk-11-using-sipml5.html >> > >> ) In a sense that sipml5 (SIP stack in the browser) is necessarily there >> on >> the browser side. My interest here is to understand the topology here how >> FS will help connecting the legacy SIP devices with browser and what will >> come into the picture in between. >> Thanks >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Webrtc-tp7580875p7586790.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/7bf03ac7/attachment.html From Sirish.MasurMohan at oa.com.au Tue Jan 29 06:47:01 2013 From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan) Date: Tue, 29 Jan 2013 14:47:01 +1100 Subject: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error References: <965759A53E43FE439E43565A7715E5F0591C085DB2@oa-exchange1.oa.com.au> Message-ID: <965759A53E43FE439E43565A7715E5F0591C085E06@oa-exchange1.oa.com.au> Hi Michael, As an experiment, I tried creating specific extensions for each phone, and then having specific dial plan, such as, With these changes, when I dial from one SNOM phone to other, the destination phone rings for a short time (say a second), after which the source phone shows a 403 forbidden error. Paste-bin of this new behaviour: http://pastebin.freeswitch.org/20515 With regards, Sirish From: Sirish Masur Mohan Sent: Tuesday, 29 January 2013 9:46 AM To: FreeSWITCH Users Help Subject: RE: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error Hi Michael, I have put the logs and SIP trace in the pastebin: http://pastebin.freeswitch.org/20514 I haven't created extensions for the SNOM phones in freeswitch, as the ACL for registration and incoming are Open - could this be a reason for the error? acl.conf.xml - sip_profiles/internal.xml With regards, Sirish From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, 26 January 2013 7:09 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error I confess that I haven't seen this specific error before. Could you get the SIP trace of this call and put it on pastebin as well? That might give you the information you need to crack the case. -MC On Thu, Jan 24, 2013 at 10:41 PM, Sirish Masur Mohan > wrote: Hello All, I think I am missing something obvious here, but I am having trouble trying to make calls between two SNOM phones. I have an open ACL for registration and I keep getting BEARERCAPABILITY_NOTAUTH error when I try to make calls. Would appreciate any inputs on how to debug this! Logs are at : http://pastebin.freeswitch.org/20504 With regards, Sirish _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/61a5682b/attachment-0001.html From jpyle at fidelityvoice.com Tue Jan 29 06:59:25 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Mon, 28 Jan 2013 22:59:25 -0500 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: I just updated from repo.profhost.eu. The most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. After disconnecting the calls the usage returned to 8.5%. I started toggling config items to see if I could impact this. I found one that seems to have an effect: rtp-timer-name in the sofia profile config. By changing it from 'soft' to 'none', the CPU utilization with 30 calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. That's great, but does it make any sense? Does an rtp-timer-name of 'none' pose any risks? - Jeff On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle wrote: > It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. > > > - Jeff > > > On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner wrote: > >> Out of curiosity does your distro use uclibc, eglibc, or glibc? >> >> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle >> wrote: >> > Hello, >> > >> > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode >> LX >> > processor (i386). I can sustain 30 concurrent calls averaging around >> 70% >> > CPU utilization by the freeswitch process, measured by top. Bypass >> media >> > and proxy media are disabled. PCMU is forced on both endpoints (no >> > transcoding). >> > >> > The problem is the RAM usage over time. The board has 256M. Idle, >> > freeswitch occupies around 4% after a fresh restart. A minute or so >> after >> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, >> 13.6%. >> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM >> usage >> > to 6-8%. >> > >> > I've not tried to troubleshoot an issue like this before. Is valgrind >> the >> > next step, or would something else make more sense? >> > >> > >> > Regards, >> > Jeff >> > >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/43bf0b5b/attachment.html From anthony.minessale at gmail.com Tue Jan 29 07:13:55 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 Jan 2013 22:13:55 -0600 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: What are the calls doing? Bridged? to an IVR? If so, which one. If you do a smaller number of calls does the same thing happen. If you can do it with 1 or 2 calls you could use valgrind. valgrind --tool=memcheck --log-file-exactly=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes /usr/local/freeswitch/bin/freeswitch On Mon, Jan 28, 2013 at 9:59 PM, Jeff Pyle wrote: > I just updated from repo.profhost.eu. The most recent timestamp on the > packages was 2013-01-28 03:41:21 GMT. Same behavior. At 5 minutes it was > using 12.5% RAM. At 40 minutes, 60.4%. After disconnecting the calls the > usage returned to 8.5%. > > I started toggling config items to see if I could impact this. I found > one that seems to have an effect: rtp-timer-name in the sofia profile > config. By changing it from 'soft' to 'none', the CPU utilization with 30 > calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. > > That's great, but does it make any sense? > > Does an rtp-timer-name of 'none' pose any risks? > > > > - Jeff > > > > On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle wrote: > >> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >> >> >> - Jeff >> >> >> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner wrote: >> >>> Out of curiosity does your distro use uclibc, eglibc, or glibc? >>> >>> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle >>> wrote: >>> > Hello, >>> > >>> > I'm running HEAD version from Jan 22 on an Alix board with an AMD >>> Geode LX >>> > processor (i386). I can sustain 30 concurrent calls averaging around >>> 70% >>> > CPU utilization by the freeswitch process, measured by top. Bypass >>> media >>> > and proxy media are disabled. PCMU is forced on both endpoints (no >>> > transcoding). >>> > >>> > The problem is the RAM usage over time. The board has 256M. Idle, >>> > freeswitch occupies around 4% after a fresh restart. A minute or so >>> after >>> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, >>> 13.6%. >>> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM >>> usage >>> > to 6-8%. >>> > >>> > I've not tried to troubleshoot an issue like this before. Is valgrind >>> the >>> > next step, or would something else make more sense? >>> > >>> > >>> > Regards, >>> > Jeff >>> >> >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/0bebd118/attachment.html From krice at freeswitch.org Tue Jan 29 07:21:14 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 28 Jan 2013 22:21:14 -0600 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: Message-ID: Ok the package timestamps/versions there don?t do us a lot of good, we need to know the version line from the FreeSWITCH CLI.. On 1/28/13 9:59 PM, "Jeff Pyle" wrote: > I just updated from?repo.profhost.eu . ?The most > recent timestamp on the packages was?2013-01-28 03:41:21 GMT. ?Same behavior. > ?At 5 minutes it was using 12.5% RAM. ?At 40 minutes, 60.4%. ?After > disconnecting the calls the usage returned to 8.5%. > > I started toggling config items to see if I could impact this. ?I found one > that seems to have an effect: ?rtp-timer-name in the sofia profile config. ?By > changing it from 'soft' to 'none', the CPU utilization with 30 calls dropped > from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. > > That's great, but does it make any sense? > > Does an rtp-timer-name of 'none' pose any risks? > > > > - Jeff > ?? > > > On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle wrote: >> It's on Voyage Linux, a cousin of Debian. ?I believe it uses glibc. >> >> >> - Jeff >> >> >> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner >> wrote: >>> Out of curiosity does your distro use uclibc, eglibc, or glibc? >>> >>> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle wrote: >>>> > Hello, >>>> > >>>> > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode >>>> LX >>>> > processor (i386). ?I can sustain 30 concurrent calls averaging around 70% >>>> > CPU utilization by the freeswitch process, measured by top. ?Bypass media >>>> > and proxy media are disabled. ?PCMU is forced on both endpoints (no >>>> > transcoding). >>>> > >>>> > The problem is the RAM usage over time. ?The board has 256M. ?Idle, >>>> > freeswitch occupies around 4% after a fresh restart. ?A minute or so >>>> after >>>> > 30 calls are nailed up the RAM usage is about 7.2%. ?After 5 minutes, >>>> 13.6%. >>>> > After 60 minutes, near 65%. ?Disconnecting the calls returns the RAM >>>> usage >>>> > to 6-8%. >>>> > >>>> > I've not tried to troubleshoot an issue like this before. ?Is valgrind >>>> the >>>> > next step, or would something else make more sense? >>>> > >>>> > >>>> > Regards, >>>> > Jeff? >>> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/957230fb/attachment-0001.html From Sirish.MasurMohan at oa.com.au Tue Jan 29 09:27:14 2013 From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan) Date: Tue, 29 Jan 2013 17:27:14 +1100 Subject: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error References: <965759A53E43FE439E43565A7715E5F0591C085DB2@oa-exchange1.oa.com.au> Message-ID: <965759A53E43FE439E43565A7715E5F0591C085E39@oa-exchange1.oa.com.au> Hi Michael, I had a 'sleep' action in my dialplan, which seemed to cause the BEARERCAPABILITY_NOTAUTH error. If I comment it out, the error goes away. However, it makes way to a new error, '[CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]' My dial plan is as follows, am I missing something? With regards, Sirish From: Sirish Masur Mohan Sent: Tuesday, 29 January 2013 2:47 PM To: 'FreeSWITCH Users Help' Subject: RE: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error Hi Michael, As an experiment, I tried creating specific extensions for each phone, and then having specific dial plan, such as, With these changes, when I dial from one SNOM phone to other, the destination phone rings for a short time (say a second), after which the source phone shows a 403 forbidden error. Paste-bin of this new behaviour: http://pastebin.freeswitch.org/20515 With regards, Sirish From: Sirish Masur Mohan Sent: Tuesday, 29 January 2013 9:46 AM To: FreeSWITCH Users Help Subject: RE: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error Hi Michael, I have put the logs and SIP trace in the pastebin: http://pastebin.freeswitch.org/20514 I haven't created extensions for the SNOM phones in freeswitch, as the ACL for registration and incoming are Open - could this be a reason for the error? acl.conf.xml - sip_profiles/internal.xml With regards, Sirish From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, 26 January 2013 7:09 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error I confess that I haven't seen this specific error before. Could you get the SIP trace of this call and put it on pastebin as well? That might give you the information you need to crack the case. -MC On Thu, Jan 24, 2013 at 10:41 PM, Sirish Masur Mohan > wrote: Hello All, I think I am missing something obvious here, but I am having trouble trying to make calls between two SNOM phones. I have an open ACL for registration and I keep getting BEARERCAPABILITY_NOTAUTH error when I try to make calls. Would appreciate any inputs on how to debug this! Logs are at : http://pastebin.freeswitch.org/20504 With regards, Sirish _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/d60ed545/attachment.html From gaz.foreman at gmail.com Tue Jan 29 00:58:15 2013 From: gaz.foreman at gmail.com (Gary Foreman) Date: Mon, 28 Jan 2013 21:58:15 +0000 Subject: [Freeswitch-users] Help with originating calls Message-ID: Hi, I'm very new to Freeswitch and having a bit of difficulty with the originate command. Essentially I'm using the CLI to originate a call however the process seems to be that the originating extension is first called, then once answered the actual call takes place. My application will take care of whether the agent is available so I want to skip the first part and have the extension dial without having to be called first, this is the way the Avaya systems I have experience with work. I've looked at the sip_auto_answer option however my sip client doesn't support auto answer. I'm just not sure on the correct way of achieving what I'm looking to do, maybe some kind of bridge would be better? Thanks in advance for your help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130128/73a39c15/attachment.html From avi at avimarcus.net Tue Jan 29 10:44:03 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 29 Jan 2013 09:44:03 +0200 Subject: [Freeswitch-users] Help with originating calls In-Reply-To: References: Message-ID: Swap the two arguments, it will call in the other order then. -Avi On Mon, Jan 28, 2013 at 11:58 PM, Gary Foreman wrote: > Hi, > > I'm very new to Freeswitch and having a bit of difficulty with the > originate command. Essentially I'm using the CLI to originate a call > however the process seems to be that the originating extension is first > called, then once answered the actual call takes place. > > My application will take care of whether the agent is available so I want > to skip the first part and have the extension dial without having to be > called first, this is the way the Avaya systems I have experience with work. > > I've looked at the sip_auto_answer option however my sip client doesn't > support auto answer. I'm just not sure on the correct way of achieving what > I'm looking to do, maybe some kind of bridge would be better? > > Thanks in advance for your help > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/d49492c5/attachment-0001.html From cstomi.levlist at gmail.com Tue Jan 29 11:36:26 2013 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Tue, 29 Jan 2013 09:36:26 +0100 Subject: [Freeswitch-users] Remove inband DTMF on record_session In-Reply-To: References: Message-ID: Hello, Anthony, could I ask your opinion, suggestions about removing inband dtmf from recorded files? We checked how we could add this feature to record_session, but I have some questions teletone dtmf detector needs more farme to detect dtmf, so I think we can't use media bug to replace the frames So we are thinking about to add a buffering and a dtmf detector if dtmf has been detected, the farmes in the buffer should be overwritten with silence It could goes to: 1) record_session app 2) file api (here buffering is implemented, but adding a detector to every file handle seems to be pretty ugly solution) 3) a new file format, that makes another level buffering Could you please let me know, if any of them could work? (I don't really like them) or advise a working one? Thanks in advance, Tamas On Fri, Jan 25, 2013 at 9:14 PM, Matt Broad wrote: > Thanks for the response Steve, is there a way of checking the dtmf type? I > have tried looking at the Sofia profile but cannot see any details > regarding dtmf. > > I assume that this is determined by my sip provider? > > > Thanks > Matt > > > On Friday, 25 January 2013, Steven Ayre wrote: > >> If you're using RFC2833 then that's not in the audio itself so won't be >> recorded. It is possible though that the caller is ending DTMF both in and >> out of band (which is fine as long as you only try to detect one or the >> other but not both). >> >> -Steve >> >> >> >> On 25 January 2013 14:27, Matt Broad wrote: >> >> Hi, >> >> sorry to interject into your message, but I was looking to do something >> similar to Tamas. >> >> I have the need to prevent the DTMF from being recorded when using >> record_session. I am using RFC2833 (i think :) ), is this at all possible >> in the current Freeswitch build? >> >> Matt >> >> >> On 21 January 2013 07:25, Tamas Jalsovszky wrote: >> >> Hello Moy, >> >> Inserting silence instead of the tone would be fine for me. I think the >> >= 50ms delay in recording is not an issue. I guess, removing from "live" >> streams would be harder and would introduce the delay you mentioned. >> So you have a plan to add such a functionality in the near future? That >> would be amazing :) >> >> Regards, >> Jalsot >> >> On Mon, Jan 21, 2013 at 2:48 AM, Moises Silva wrote: >> >> On Sun, Jan 20, 2013 at 4:12 PM, Tamas Jalsovszky wrote: >> >> Hello, >> >> Is there a way to to remove inbad DTMF from the recorded wav file >> (conditionaly)? >> If not, how could be that done the easyiest way and a way it could get >> into FS git? >> >> As FS has a great DTMF detector and it has probably all the stuff doing >> such a feature, I think it would be much better than using an external tool >> for removing DTMF from finished wav file - and it would consume more I/O >> probably. >> >> >> Removing it would not be that hard (ie, inserting silence instead). >> Removing it without leaving a trace any DTMF was ever there sounds a tad >> more difficult. If you're ok with inserting a silence instead of the tone, >> extending the record_session app to detect and remove DTMF is probably what >> I'd do. >> >> It is also possible to create a new app that monitors the audio stream >> and silence the stream when DTMF is detected, but since the detection takes >> a few milliseconds (~50ms), there would be a bit of "bleeding" of DTMF in >> the signal, unless the new app also delays the signal by 50ms and then has >> a way to strip the DTMF completely. >> >> *Moises Silva >> **Manager, Software Engineering*** >> >> msilva at sangoma.com >> >> Sangoma Technologies >> >> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >> >> > > -- > Thanks > Matt > > This email and any attachments to it are confidential and are intended > solely for the use of the individual to whom it is addressed. Any views or > opinions expressed are solely those of the author and do not necessarily > represent those of InverOak Limited. > > If you are not the intended recipient of this email, you must neither take > any action based upon its contents, nor copy or show it to anyone. Please > contact the sender if you believe you have received this email in error. > > This email including any attachments cannot be guaranteed to be 100% > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, out-dated, or containing viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message which arise as a result of email transmission. > > InverOak Limited is a company registered in England & Wales under company > number 04529594, whose registered address is Old Barn house, 2 Wannions > Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/998a53f7/attachment.html From alessio at asistar.it Tue Jan 29 12:05:13 2013 From: alessio at asistar.it (Alessio) Date: Tue, 29 Jan 2013 10:05:13 +0100 Subject: [Freeswitch-users] Patton FXO configuration on Freeswitch - more details In-Reply-To: References: <5102AAF6.2070201@asistar.it> <51062C9C.8020701@asistar.it> Message-ID: <510790C9.204@asistar.it> Nella citazione in data luned? 28 gennaio 2013 19:43:12, Michael Collins ha scritto: > > Hi Michael, > thanks for the reply. > > So if I register in directory as a phone how do I get out of the > gateway, for example through a bridge? > > Alessio > > > User directory is for inbound registrations and SIP auth challenges > for devices connecting to the CudaTel. Gateways are the opposite - > they are for auth challenges and registrations going from the CudaTel > to other devices. To dial out a gateway you just use the syntax: > > data="sofia/gateway/gateway_name/${phone_number_to_send_to_gateway}"/> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org So for incoming calls and recording to the PBX should I configure FXO in conf/directory. To make outgoing calls through the FXO I configure a gateway in conf/sip_profile. Is that correct? If so, how do I configure the gateway? Thank you, Alessio From jaasmailing at gmail.com Tue Jan 29 13:18:34 2013 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Tue, 29 Jan 2013 11:18:34 +0100 Subject: [Freeswitch-users] Kill lua script from CLI Message-ID: <5107A1FA.5020904@gmail.com> Hi all, is there any way to kill a lua script from FS CLI? I have some scripts that loop internally and I want to stop them. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/d0d0cc47/attachment.html From mehroz.ashraf85 at gmail.com Tue Jan 29 14:14:11 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Tue, 29 Jan 2013 03:14:11 -0800 (PST) Subject: [Freeswitch-users] Direcotry Status with mod_xml_curl In-Reply-To: References: <1359375027857-7586787.post@n2.nabble.com> Message-ID: <1359458051068-7586828.post@n2.nabble.com> Thanks guys. I guess using mod_sofia's odbc-dsn param is what helped me out. ODBC using MySQL engine to store registration information in DB (that was already created when settting up xml_curl DB). It already had tables like "Sip_registration" and "SIP_presence" and some others too. However, this is not exactly what asterisk-realtime is about :) as i had to play with ODBC, Mysql, configuring 4-5 XML files, but surely a lot helpful if you want to query that user is online or not (YES, if the record exists). For those still lost in solution, this post is also helpful http://freeswitch-users.2379917.n2.nabble.com/Clustering-storing-registration-server-IP-in-DB-td6375234.html#a6375434 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Direcotry-Status-with-mod-xml-curl-tp7586787p7586828.html Sent from the freeswitch-users mailing list archive at Nabble.com. From cal.leeming at simplicitymedialtd.co.uk Tue Jan 29 15:00:31 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 29 Jan 2013 12:00:31 +0000 Subject: [Freeswitch-users] Direcotry Status with mod_xml_curl In-Reply-To: References: <1359375027857-7586787.post@n2.nabble.com> Message-ID: Ah I didn't know about this - thanks :) Cal On Mon, Jan 28, 2013 at 5:45 PM, Steven Ayre wrote: > It's also possible to store the information in a ODBC database > (core-db-dsn) and mod_sofia's odbc-dsn profile param. > > You can then query (read-only!) the registration DB tables directory. > > > If you want to interact in realtime (ie no polling) collecting events and > running async api commands via ESL is the way to go. It's similar to > Asterisk's AMI and the events are very powerful. > > -Steve > > > > On 28 January 2013 17:07, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello, >> >> You want to use mod_event_socket to retrieve the registration status of a >> user. >> >> See the following; >> http://wiki.freeswitch.org/wiki/Event_Socket >> >> Hope this helps. >> >> Cal >> >> >> On Mon, Jan 28, 2013 at 12:10 PM, mehroz wrote: >> >>> Hi, >>> >>> I have been working with asterisk previously, recently came over >>> Freeswitch >>> platform. >>> >>> I dig up for the similar fucntionality of asteriks-realtime and found >>> MOD_XML_CURL as the nearest match for dynamic directory(extension) >>> creation. >>> One good thing about asterisk-realtime was, it was compact in a single >>> table >>> and direct conversation with Asterisk and DB is possible , whereas in >>> Freeswitch its through Webserver and curl requests. Watever, i accepted >>> that! but one more thing i am missing is the status of SIP Registration >>> (registered or not). >>> >>> Again, in asterisk realtime i can see the RTP port assigned to extension >>> when it registers, this way i can see the current status reading the DB , >>> and port number. >>> >>> Now the question is , how can i see this with mod_xml_curl ? and also, if >>> any one can help me with the above mentioned grudges and crushes! >>> >>> >>> >>> -- >>> View this message in context: >>> http://freeswitch-users.2379917.n2.nabble.com/Direcotry-Status-with-mod-xml-curl-tp7586787.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/f8c9f3af/attachment-0001.html From vbvbrj at gmail.com Tue Jan 29 16:22:37 2013 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 29 Jan 2013 15:22:37 +0200 Subject: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 In-Reply-To: References: <50F909AD.2010308@gmail.com> <50F91490.7080006@gmail.com> Message-ID: <5107CD1D.9020302@gmail.com> On 18.01.2013 18:19, Steven Ayre wrote: > First of all update to the latest version of v1.2.stable or master - it > may be something that's already fixed if you're on an old version. Updated to FreeSWITCH Version 1.3.13b+git~20130118T050605Z~cad607d72e (git cad607d 2013-01-18 05:06:05Z) After a many days test it doesn't generate this segfault anymore. But again in case of coredumps, how to specify where to save this files? As they does not have permission to fs binary folder. -- Mimiko desu. From nbhatti at gmail.com Tue Jan 29 17:17:09 2013 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 29 Jan 2013 17:17:09 +0300 Subject: [Freeswitch-users] continue_on_fail does not works with limit? Message-ID: <7D835399-B84F-4D1C-8CB4-64E8EDB0BC2C@gmail.com> I am using limit_execute to bridge a call. My continue_on_fail is set to false since I don't want to try the next gateway if the call is not tried because there is 1 call already first gateway. Since the call is never tried with the first gateway because of limit, there is no FAILURE from the gateway for continue_on_fail to react. Is there a way to make limit work with continue_on_fail? Or is there a better way to send multiple gateways in the dial plan where I only want to try the next gateway in the list if first gateway's limit it reached? Thanks, -- Muhammad Naseer Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/118e59f8/attachment.html From andrew at cassidywebservices.co.uk Tue Jan 29 17:31:56 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 29 Jan 2013 14:31:56 +0000 Subject: [Freeswitch-users] Postgres in the Core Message-ID: So, I've got postgres set up in the core as my main database. But what does this mean for other modules that each allow you to specify their own odbc-dns options, such as sofia, callcenter, voicemail, etc? -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/0568924a/attachment.html From jpyle at fidelityvoice.com Tue Jan 29 17:33:48 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Tue, 29 Jan 2013 09:33:48 -0500 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: Version: FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 (-n20130122T122521Z-1~squeeze+1) The calls are bridged, from one sofia profile to another. Unfortunately two concurrent calls doesn't seem to trigger the same behavior. - Jeff On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: > Ok the package timestamps/versions there don?t do us a lot of good, we > need to know the version line from the FreeSWITCH CLI.. > > > On 1/28/13 9:59 PM, "Jeff Pyle" wrote: > > I just updated from repo.profhost.eu . The > most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same > behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. > After disconnecting the calls the usage returned to 8.5%. > > > I started toggling config items to see if I could impact this. I found > one that seems to have an effect: rtp-timer-name in the sofia profile > config. By changing it from 'soft' to 'none', the CPU utilization with 30 > calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. > > That's great, but does it make any sense? > > Does an rtp-timer-name of 'none' pose any risks? > > > > - Jeff > > > > On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle > wrote: > > It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. > > > - Jeff > > > On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner > wrote: > > Out of curiosity does your distro use uclibc, eglibc, or glibc? > > On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle > wrote: > > Hello, > > > > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode > LX > > processor (i386). I can sustain 30 concurrent calls averaging around 70% > > CPU utilization by the freeswitch process, measured by top. Bypass media > > and proxy media are disabled. PCMU is forced on both endpoints (no > > transcoding). > > > > The problem is the RAM usage over time. The board has 256M. Idle, > > freeswitch occupies around 4% after a fresh restart. A minute or so > after > > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, > 13.6%. > > After 60 minutes, near 65%. Disconnecting the calls returns the RAM > usage > > to 6-8%. > > > > I've not tried to troubleshoot an issue like this before. Is valgrind > the > > next step, or would something else make more sense? > > > > > > Regards, > > Jeff > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/99341d27/attachment.html From steveayre at gmail.com Tue Jan 29 17:52:04 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 29 Jan 2013 14:52:04 +0000 Subject: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 In-Reply-To: <5107CD1D.9020302@gmail.com> References: <50F909AD.2010308@gmail.com> <50F91490.7080006@gmail.com> <5107CD1D.9020302@gmail.com> Message-ID: Coredumps are generated by the operating system, not FreeSWITCH. See 'man 5 core'. It is configurable, to an extent. The default is a file named 'core' in the current working directory of the process. You can adjust that behaviour through procfs using /proc/sys/kernel/core_pattern. It can include / characters to set a directory. Note it's a global setting. For example: $ echo '/tmp/core.%e.%p' > /proc/sys/kernel/core_pattern That would result in coredump files being created such as /tmp/core.freeswitch.23523 (example). You'll want to pick a suitable directory with enough space and suitable permissions. -Steve On 29 January 2013 13:22, Mimiko wrote: > On 18.01.2013 18:19, Steven Ayre wrote: > > First of all update to the latest version of v1.2.stable or master - it > > may be something that's already fixed if you're on an old version. > > > Updated to FreeSWITCH Version 1.3.13b+git~20130118T050605Z~cad607d72e > (git cad607d 2013-01-18 05:06:05Z) > > After a many days test it doesn't generate this segfault anymore. > > But again in case of coredumps, how to specify where to save this files? > As they does not have permission to fs binary folder. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/f83ca37f/attachment-0001.html From steveayre at gmail.com Tue Jan 29 18:14:14 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 29 Jan 2013 15:14:14 +0000 Subject: [Freeswitch-users] Kill lua script from CLI In-Reply-To: <5107A1FA.5020904@gmail.com> References: <5107A1FA.5020904@gmail.com> Message-ID: No. LUA doesn't have any job control right now. Any scripts you have running should be looping checking a condition. For scripts run on calls only loop while session:ready(), you can then hangup the call (eg from cli) and that would break the loop and end the script. For ones started at startup you need to have them check for a stop event, there are examples around. -Steve On 29 January 2013 10:18, Carlo Dimaggio wrote: > Hi all, > > is there any way to kill a lua script from FS CLI? > I have some scripts that loop internally and I want to stop them. > > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/c555ef15/attachment.html From nbhatti at gmail.com Tue Jan 29 18:27:23 2013 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 29 Jan 2013 18:27:23 +0300 Subject: [Freeswitch-users] continue_on_fail does not works with limit? In-Reply-To: <7D835399-B84F-4D1C-8CB4-64E8EDB0BC2C@gmail.com> References: <7D835399-B84F-4D1C-8CB4-64E8EDB0BC2C@gmail.com> Message-ID: <2803C603-1B88-4240-BF3A-E0281607CA6C@gmail.com> Sorry bad post .. ignore it :( Thanks, -- Muhammad Naseer Bhatti On Jan 29, 2013, at 5:17 PM, Muhammad Naseer Bhatti wrote: > > I am using limit_execute to bridge a call. > > > > > > My continue_on_fail is set to false since I don't want to try the next gateway if the call is not tried because there is 1 call already first gateway. Since the call is never tried with the first gateway because of limit, there is no FAILURE from the gateway for continue_on_fail to react. Is there a way to make limit work with continue_on_fail? Or is there a better way to send multiple gateways in the dial plan where I only want to try the next gateway in the list if first gateway's limit it reached? > > > Thanks, > -- > Muhammad Naseer Bhatti > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/46011793/attachment.html From grcamauer at gmail.com Tue Jan 29 19:03:53 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 29 Jan 2013 13:03:53 -0300 Subject: [Freeswitch-users] fs_encode and segmentation fault In-Reply-To: References: Message-ID: I am running FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z) How do I tell if this is Master or Stable? How do I switch between the two? I will try to collect the backtrace and post on Jira. Thanks On Mon, Jan 28, 2013 at 9:57 PM, Steven Ayre wrote: > Can you reproduce this on Git master head? > > Please enable coredumps (ulimit -c unlimited) and collect a backtrace, > then open a Jira bug ticket and attach the backtrace. > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 > http://jira.freeswitch.org/ > > -Steve > > > > On 28 January 2013 23:30, Guillermo Ruiz Camauer wrote: > >> >> >> I am trying the following command: >> fs_encode -l mod_sangoma_codec greeting.wav greeting.G729 >> >> which returns: >> >> 31663 Segmentation fault >> >> If I try to convert to PCMU it works fine: >> fs_encode -l mod_sangoma_codec greeting.wav greeting.PCMU >> >> Making and receiving calls with G729 works fine. >> >> A similar command on a machine with a DIGIUM card works fine: >> fs_encode -l mod_dahdi_codec greeting.wav greeting.G729 >> >> The only difference I see is the card and the module being loaded. Any >> idea of what could be happenning? Do I need to load some other module for >> the Sangoma card? >> >> Thanks in advance >> >> -- >> Guillermo >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/5d517e30/attachment.html From krice at freeswitch.org Tue Jan 29 19:18:25 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 29 Jan 2013 10:18:25 -0600 Subject: [Freeswitch-users] fs_encode and segmentation fault In-Reply-To: Message-ID: 1.X.... Where X is even that?s a stable branch, where X is odd that?s a dev branch, currently those are 1.2 and 1.3 .... ?git status? will also show you from your source tree checked out from git which branch you are on... To change between the branches you do the following (from your source tree, adjusting paths for your install as required, I will assume you have installed into /usr/local/freeswitch) # cd /usr/src/freeswitch ;#(or wherever that source is) # git clean ?fdx # rm ?rf /usr/local/freeswich/{mod,bin,lib}/* ;#Adjust path if needed # git checkout BRANCH ;# where branch name is v1.2.stable or master # git pull ;#make sure you are up to date on this branch Now starting at bootstrap.sh install as normal, this will not overwrite your configs, and yes it is necessary to start over clean, if you do not it is possible that there are patches in the libs that are not picked up and recompliled leaving you with branch to branch build skew which will result in all sorts of weird behavior. Also please be sure that any bugs you file are against the master branch as they will get fixed there first then rolled down to the stable branch. K On 1/29/13 10:03 AM, "Guillermo Ruiz Camauer" wrote: > I am running?FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git > e04eab7 2012-12-29 00:17:59Z) > > How do I tell if this is Master or Stable? ?How do I switch between the two? > > I will try to collect the backtrace and post on Jira. > > Thanks > > > On Mon, Jan 28, 2013 at 9:57 PM, Steven Ayre wrote: >> Can you reproduce this on Git master head? >> >> Please enable coredumps (ulimit -c unlimited) and collect a backtrace, then >> open a Jira bug ticket and attach the backtrace. >> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_ >> .28Linux.2FUnix.29 >> http://jira.freeswitch.org/ >> >> -Steve >> >> >> >> On 28 January 2013 23:30, Guillermo Ruiz Camauer wrote: >>> >>> >>> I am trying the following command: >>> fs_encode -l mod_sangoma_codec greeting.wav greeting.G729 >>> >>> which returns: >>> >>> 31663 Segmentation fault >>> >>> If I try to convert to PCMU it works fine: >>> fs_encode -l mod_sangoma_codec greeting.wav greeting.PCMU >>> >>> Making and receiving calls with G729 works fine. >>> >>> A similar command on a machine with a DIGIUM card works fine: >>> fs_encode -l mod_dahdi_codec greeting.wav greeting.G729 >>> >>> The only difference I see is the card and the module being loaded. Any idea >>> of what could be happenning? ?Do I need to load some other module for the >>> Sangoma card? >>> >>> Thanks in advance -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/f16c451b/attachment-0001.html From kkgp20 at gmail.com Tue Jan 29 15:05:31 2013 From: kkgp20 at gmail.com (kkgp20 at gmail.com) Date: Tue, 29 Jan 2013 13:05:31 +0100 Subject: [Freeswitch-users] [mod_sofia] Is the SIP Presence (SIMPLE) functionality limited to 2 states on FreeSWITCH? Message-ID: <002401cdfe18$f04b9d90$d0e2d8b0$@gmail.com> Good morning, I am trying to understand and make working the SIP Presence functionality (SIMPLE) with the FreeSWITCH server. I have noticed that FreeSWITCH seems to support only two states: registered and not registered (online, offline). I have installed two SIP Clients: PortGO and Zoiper Communicator, both have to functionality to set a user status like: busy, away, in a call, on the lunch etc. By the Wireshark software I have seen that changing the user status on PortGO does not cause to send any SIP NOTIFY or PUBLISH messages - which seems to be a bug, but the Zoiper was doing it. So then I have noticed that when I have changed the user state on one of the clients, the second client all the time was receiving the NOTIFY message from FreeSWITCH with "Available" status (no matter which state I have chosen, despite offline). Only in case when I have switched to offline, the client have received "Not registered" message. My question is if it is possible to configure the FreeSWITCH server to support other states? It means any text which is going inside the XML tag? Because now it seems that this tag is ignored and FreeSWITCH is sending the Available. Thank you for help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/d5bb448d/attachment.html From anthony.minessale at gmail.com Tue Jan 29 20:04:34 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Jan 2013 11:04:34 -0600 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: Can you systematically increase the current call count and see where you do see something? On a system with limited ram you can also consider stripping all the .so files in the mods and lib dir but you will need to put non-stripped ones in for any debugging. I don't really see a correlation on how not using a timer could trigger an sustained memory increase so that's why I'd like you to step up the number and see if you can find a number of calls that tops out because usually there always is a magic number where it will hover and go up and down a meg at a time. SIP calls are required to keep state data around for at least 30 seconds after a call ends and there are a number of pools in the code that inflate once and do not return the memory. Its usually possible to identify the high watermark on a particular box. For instance the machine we host the conference call on launches using 25 megs and hovers at about 350 megs once it has accumulated all the pool memory it needs over time. On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle wrote: > Version: > FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 > (-n20130122T122521Z-1~squeeze+1) > > The calls are bridged, from one sofia profile to another. > > Unfortunately two concurrent calls doesn't seem to trigger the same > behavior. > > > - Jeff > > > On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: > >> Ok the package timestamps/versions there don?t do us a lot of good, we >> need to know the version line from the FreeSWITCH CLI.. >> >> >> On 1/28/13 9:59 PM, "Jeff Pyle" wrote: >> >> I just updated from repo.profhost.eu . The >> most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same >> behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. >> After disconnecting the calls the usage returned to 8.5%. >> >> >> I started toggling config items to see if I could impact this. I found >> one that seems to have an effect: rtp-timer-name in the sofia profile >> config. By changing it from 'soft' to 'none', the CPU utilization with 30 >> calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. >> >> That's great, but does it make any sense? >> >> Does an rtp-timer-name of 'none' pose any risks? >> >> >> >> - Jeff >> >> >> >> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle >> wrote: >> >> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >> >> >> - Jeff >> >> >> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner >> wrote: >> >> Out of curiosity does your distro use uclibc, eglibc, or glibc? >> >> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle >> wrote: >> > Hello, >> > >> > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode >> LX >> > processor (i386). I can sustain 30 concurrent calls averaging around >> 70% >> > CPU utilization by the freeswitch process, measured by top. Bypass >> media >> > and proxy media are disabled. PCMU is forced on both endpoints (no >> > transcoding). >> > >> > The problem is the RAM usage over time. The board has 256M. Idle, >> > freeswitch occupies around 4% after a fresh restart. A minute or so >> after >> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, >> 13.6%. >> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM >> usage >> > to 6-8%. >> > >> > I've not tried to troubleshoot an issue like this before. Is valgrind >> the >> > next step, or would something else make more sense? >> > >> > >> > Regards, >> > Jeff >> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/c6e02dd5/attachment.html From anthony.minessale at gmail.com Tue Jan 29 20:07:11 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Jan 2013 11:07:11 -0600 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: References: Message-ID: The best idea is to set a global variable in vars.xml with the dsn and dist it down to those modules. Unless you don't need it to be pg, then you can leave them on sqlite. On Tue, Jan 29, 2013 at 8:31 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > So, I've got postgres set up in the core as my main database. > > But what does this mean for other modules that each allow you to specify > their own odbc-dns options, such as sofia, callcenter, voicemail, etc? > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/20e80899/attachment.html From anthony.minessale at gmail.com Tue Jan 29 20:41:58 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Jan 2013 11:41:58 -0600 Subject: [Freeswitch-users] Kill lua script from CLI In-Reply-To: References: <5107A1FA.5020904@gmail.com> Message-ID: Here is a trick you can do that kind of created itself through evolution. Use the loopback channel to execute the script and name it a job name with a variable // Enter this from the cli (depending on where you call it from you may not need to escape the commas but // from the cli you will for sure. originate loopback/set:job=1234\,answer\,park/default/inline lua:loop.lua inline switch_ivr.c:1801 Transfer loopback/set:job=loop.lua,answer,park-a to inline[lua:loop.lua at default] // now your script is running in the dummy channel and you can hang it up whenever you want by killing the uuid or calling "hupall manager_request job loop.lua" hupall manager_request job loop.lua +OK hangup all channels matching [job]=[loop.lua] with cause: MANAGER_REQUEST On Tue, Jan 29, 2013 at 9:14 AM, Steven Ayre wrote: > No. LUA doesn't have any job control right now. > > Any scripts you have running should be looping checking a condition. For > scripts run on calls only loop while session:ready(), you can then hangup > the call (eg from cli) and that would break the loop and end the script. > > For ones started at startup you need to have them check for a stop event, > there are examples around. > > -Steve > > > > On 29 January 2013 10:18, Carlo Dimaggio wrote: > >> Hi all, >> >> is there any way to kill a lua script from FS CLI? >> I have some scripts that loop internally and I want to stop them. >> >> >> Regards, >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/6d47c73f/attachment-0001.html From grcamauer at gmail.com Tue Jan 29 21:42:58 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 29 Jan 2013 15:42:58 -0300 Subject: [Freeswitch-users] fs_encode and segmentation fault In-Reply-To: References: Message-ID: Thanks for the detailed reply. I have compiled MASTER and reproduced the same results. I opened a JIRA ticket against the original version. I am updating it now to post the findings against MASTER. On Tue, Jan 29, 2013 at 1:18 PM, Ken Rice wrote: > 1.X.... Where X is even that?s a stable branch, where X is odd that?s a > dev branch, currently those are 1.2 and 1.3 .... > > ?git status? will also show you from your source tree checked out from git > which branch you are on... > > To change between the branches you do the following (from your source > tree, adjusting paths for your install as required, I will assume you have > installed into /usr/local/freeswitch) > > # cd /usr/src/freeswitch ;#(or wherever that source is) > # git clean ?fdx > # rm ?rf /usr/local/freeswich/{mod,bin,lib}/* ;#Adjust path if needed > # git checkout BRANCH ;# where branch name is v1.2.stable or master > # git pull ;#make sure you are up to date on this branch > > Now starting at bootstrap.sh install as normal, this will not overwrite > your configs, and yes it is necessary to start over clean, if you do not it > is possible that there are patches in the libs that are not picked up and > recompliled leaving you with branch to branch build skew which will result > in all sorts of weird behavior. > > Also please be sure that any bugs you file are against the master branch > as they will get fixed there first then rolled down to the stable branch. > > K > > > > > On 1/29/13 10:03 AM, "Guillermo Ruiz Camauer" wrote: > > I am running FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 > (git e04eab7 2012-12-29 00:17:59Z) > > How do I tell if this is Master or Stable? How do I switch between the > two? > > I will try to collect the backtrace and post on Jira. > > Thanks > > > On Mon, Jan 28, 2013 at 9:57 PM, Steven Ayre wrote: > > Can you reproduce this on Git master head? > > Please enable coredumps (ulimit -c unlimited) and collect a backtrace, > then open a Jira bug ticket and attach the backtrace. > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Creating_A_Backtrace_With_gdb_.28Linux.2FUnix.29 > http://jira.freeswitch.org/ > > -Steve > > > > On 28 January 2013 23:30, Guillermo Ruiz Camauer > wrote: > > > > I am trying the following command: > fs_encode -l mod_sangoma_codec greeting.wav greeting.G729 > > which returns: > > 31663 Segmentation fault > > If I try to convert to PCMU it works fine: > fs_encode -l mod_sangoma_codec greeting.wav greeting.PCMU > > Making and receiving calls with G729 works fine. > > A similar command on a machine with a DIGIUM card works fine: > fs_encode -l mod_dahdi_codec greeting.wav greeting.G729 > > The only difference I see is the card and the module being loaded. Any > idea of what could be happenning? Do I need to load some other module for > the Sangoma card? > > Thanks in advance > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/bab428a6/attachment.html From avi at avimarcus.net Tue Jan 29 21:57:36 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 29 Jan 2013 20:57:36 +0200 Subject: [Freeswitch-users] SPA-2102 Gives 486 instead of Call Waiting In-Reply-To: References: Message-ID: Any suggestions on how to figure this out? I'd really appreciate it! -Avi On Mon, Jan 7, 2013 at 3:12 PM, Avi Marcus wrote: > Here's an intersting one that seems to have no results on google or in > debug info: > > I have an SPA-2102 (firmware 5.2.10) that returns 486/Busy Here when a > call comes in to the active line. > > "call waiting" is turned on, so it should be beeping in instead... > > I got these logs off the SPA but it doesn't seem very helpful: (and there > should be more, I think) > > ==> /var/log/syslog <== > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) > Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 > Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 100 Trying > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 486 Busy Here > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 > Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) > Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) > Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 > Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 > > > I saw one suggestion to do a factory reset that fixed the problem, but > that's a bit troublesome at the moment since the client can't > load provisioning files (dumb network.) > > -Avi Marcus > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/c6f0cc76/attachment.html From mario_fs at mgtech.com Tue Jan 29 21:58:34 2013 From: mario_fs at mgtech.com (Mario G) Date: Tue, 29 Jan 2013 10:58:34 -0800 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: Probably not be related, but you never know: on OSX since Nov/Dec there has been a memory leak on 1.2 and head that occurs for in/out/and registrations. I had to triple memory and recycle FreeSwitch every 2-3 days since then. Will open a Jira when I can obtain detailed info and possibly run valgrind. I am also also waiting to update the main FS computer from 10.6.8 to 10.8.3 to see its effect. Hopefully in Feb. Mario G On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: > Can you systematically increase the current call count and see where you do see something? > On a system with limited ram you can also consider stripping all the .so files in the mods and lib dir but you will need to put non-stripped ones in for any debugging. > > I don't really see a correlation on how not using a timer could trigger an sustained memory increase so that's why I'd like you to step up the number and see if you can find a number of calls that tops out because usually there always is a magic number where it will hover and go up and down a meg at a time. > > SIP calls are required to keep state data around for at least 30 seconds after a call ends and there are a number of pools in the code that inflate once and do not return the memory. Its usually possible to identify the high watermark on a particular box. > > For instance the machine we host the conference call on launches using 25 megs and hovers at about 350 megs once it has accumulated all the pool memory it needs over time. > > > > > > On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle wrote: > Version: > FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 (-n20130122T122521Z-1~squeeze+1) > > The calls are bridged, from one sofia profile to another. > > Unfortunately two concurrent calls doesn't seem to trigger the same behavior. > > > - Jeff > > > On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: > Ok the package timestamps/versions there don?t do us a lot of good, we need to know the version line from the FreeSWITCH CLI.. > > > On 1/28/13 9:59 PM, "Jeff Pyle" wrote: > > I just updated from repo.profhost.eu . The most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. After disconnecting the calls the usage returned to 8.5%. > > > I started toggling config items to see if I could impact this. I found one that seems to have an effect: rtp-timer-name in the sofia profile config. By changing it from 'soft' to 'none', the CPU utilization with 30 calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. > > That's great, but does it make any sense? > > Does an rtp-timer-name of 'none' pose any risks? > > > > - Jeff > > > > On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle wrote: > It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. > > > - Jeff > > > On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner wrote: > Out of curiosity does your distro use uclibc, eglibc, or glibc? > > On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle wrote: > > Hello, > > > > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode LX > > processor (i386). I can sustain 30 concurrent calls averaging around 70% > > CPU utilization by the freeswitch process, measured by top. Bypass media > > and proxy media are disabled. PCMU is forced on both endpoints (no > > transcoding). > > > > The problem is the RAM usage over time. The board has 256M. Idle, > > freeswitch occupies around 4% after a fresh restart. A minute or so after > > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, 13.6%. > > After 60 minutes, near 65%. Disconnecting the calls returns the RAM usage > > to 6-8%. > > > > I've not tried to troubleshoot an issue like this before. Is valgrind the > > next step, or would something else make more sense? > > > > > > Regards, > > Jeff > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/35e38a8b/attachment-0001.html From ga at steadfasttelecom.com Tue Jan 29 22:08:02 2013 From: ga at steadfasttelecom.com (Gilad Abada) Date: Tue, 29 Jan 2013 14:08:02 -0500 Subject: [Freeswitch-users] SPA-2102 Gives 486 instead of Call Waiting In-Reply-To: References: Message-ID: <3010184978485284118@unknownmsgid> Avi, You can try to enable syslog trough the GUI and see what the spa 2102 os actually doing. I think it's under the system tab you have syslog and debug server. Sent from my mobile device. On Jan 29, 2013, at 2:05 PM, Avi Marcus wrote: Any suggestions on how to figure this out? I'd really appreciate it! -Avi On Mon, Jan 7, 2013 at 3:12 PM, Avi Marcus wrote: > Here's an intersting one that seems to have no results on google or in > debug info: > > I have an SPA-2102 (firmware 5.2.10) that returns 486/Busy Here when a > call comes in to the active line. > > "call waiting" is turned on, so it should be beeping in instead... > > I got these logs off the SPA but it doesn't seem very helpful: (and there > should be more, I think) > > ==> /var/log/syslog <== > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) > Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 > Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 100 Trying > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 486 Busy Here > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 > Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 > Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) > Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) > Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 > Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 > > > I saw one suggestion to do a factory reset that fixed the problem, but > that's a bit troublesome at the moment since the client can't > load provisioning files (dumb network.) > > -Avi Marcus > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/2ef1f52c/attachment.html From avi at avimarcus.net Tue Jan 29 22:24:58 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 29 Jan 2013 21:24:58 +0200 Subject: [Freeswitch-users] SPA-2102 Gives 486 instead of Call Waiting In-Reply-To: <3010184978485284118@unknownmsgid> References: <3010184978485284118@unknownmsgid> Message-ID: I did, that's how I got the log in my first email... Was I supposed to find more than that? -Avi On Tue, Jan 29, 2013 at 9:08 PM, Gilad Abada wrote: > Avi, > > You can try to enable syslog trough the GUI and see what the spa 2102 os > actually doing. I think it's under the system tab you have syslog and debug > server. > > Sent from my mobile device. > > On Jan 29, 2013, at 2:05 PM, Avi Marcus wrote: > > Any suggestions on how to figure this out? I'd really appreciate it! > > -Avi > > > On Mon, Jan 7, 2013 at 3:12 PM, Avi Marcus wrote: > >> Here's an intersting one that seems to have no results on google or in >> debug info: >> >> I have an SPA-2102 (firmware 5.2.10) that returns 486/Busy Here when a >> call comes in to the active line. >> >> "call waiting" is turned on, so it should be beeping in instead... >> >> I got these logs off the SPA but it doesn't seem very helpful: (and there >> should be more, I think) >> >> ==> /var/log/syslog <== >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 100 Trying >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 486 Busy Here >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >> >> >> I saw one suggestion to do a factory reset that fixed the problem, but >> that's a bit troublesome at the moment since the client can't >> load provisioning files (dumb network.) >> >> -Avi Marcus >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/b4c639a0/attachment.html From mario_fs at mgtech.com Tue Jan 29 22:31:43 2013 From: mario_fs at mgtech.com (Mario G) Date: Tue, 29 Jan 2013 11:31:43 -0800 Subject: [Freeswitch-users] adding In-Reply-To in sip header not working In-Reply-To: References: Message-ID: <0F4F8E19-73FC-460F-A5AB-B7A2CF8CD78B@mgtech.com> Thanks, I tried both suggestions but no love. When I used sip_h_In-Reply-To=${sip_call_id} used as below, the trace showed all normal but the cell phone does not ring at all. When I removed everything the cell rang but the original number was not passed. BTW, the sip_call_id was translated to (#s altered): sip_h_In-Reply-To=3912345-9123456295-612341 at msw1.telengy.net, could that be an issue with ATT not liking it? Mario G On Jan 27, 2013, at 3:02 PM, Richard Brady wrote: > Ok, nifty. They are letting you present a number you do not own as Caller ID on an outbound call if that outbound call is a forwarded leg of an inbound call. > > They do this by looking the In-Reply-To header of the INVITE for the forwarded leg, which should contain the Call-ID of the orignal leg. > > So you need to copy the Call-ID in order to authorize the Caller ID. > > A couple things: > > 1. From the docs: effective_caller_id_name Sets the effective callerid name. This is automatically exported to the B-leg; however, it is not valid in an origination string. In other words, set this before calling bridge, otherwise use origination_caller_id_name > > 2. You shouldn't care about 1 above as it should be copied across from the A leg by default and you are not modifying it, so remove effective_caller_id_name and don't bother with origination_caller_id_name either. > > 3. You should use sip_h_ not sip_rh_ because you want the header in the new INVITE going out. > > Perhaps try: > > > > Hope this helps. > > Richard > > > On 21 January 2013 19:54, Mario G wrote: > Thanks, apparently I had it wrong, the doc below states that the PBX must support it incoming, they pointed me to using effective_caller_id which I added to the bridge but it still does not work. Would love to fix this since the cell phones currently have no idea who is calling. > Mario G > > > > Please note that this feature is ONLY AVAILABLE for customers using a SIP PBX that either supports (or allows the configuring of) the "in-reply-to" header (defined by RFC 3261) for incoming calls which are forwarded to an outbound trunk. In these instances Callcentric will "Pass-Through" the CallerID from the original call which was received to the outbound bridged/forwarded call. > > On Jan 19, 2013, at 5:13 PM, Richard Brady wrote: > >> On 20 January 2013 00:06, Mario G wrote: >> I never did this so I must be missing something, I tried both below but the bridge then fails. Can anyone shed some light on what I am doing wrong. My ITSP now supports in-reply-to so I can pass the caller ID to a forwarded call from FS. >> >> In-Reply-To should contain a Call-ID not a caller ID. They are very different. >> >> The following would make a bit more sense, but still not a lot: >> >> >> >> Using In-Reply-To in a response doesn't seem right to me. I would expect it to appear in an INVITE. For for example, you get a missed call and you call the person back, then the INVITE for the callback would have a new Call-ID but the original Call-ID in the In-Reply-To header. That said, I have no idea what your ITSPs intended use for the header is. >> >> Richard >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/d569fa9c/attachment-0001.html From ga at steadfasttelecom.com Tue Jan 29 22:34:47 2013 From: ga at steadfasttelecom.com (Gilad Abada) Date: Tue, 29 Jan 2013 14:34:47 -0500 Subject: [Freeswitch-users] SPA-2102 Gives 486 instead of Call Waiting In-Reply-To: References: <3010184978485284118@unknownmsgid> Message-ID: <4522496924514533582@unknownmsgid> I think if you turn debug and syslog on level 3 it gives more info Sent from my mobile device. On Jan 29, 2013, at 2:28 PM, Avi Marcus wrote: I did, that's how I got the log in my first email... Was I supposed to find more than that? -Avi On Tue, Jan 29, 2013 at 9:08 PM, Gilad Abada wrote: > Avi, > > You can try to enable syslog trough the GUI and see what the spa 2102 os > actually doing. I think it's under the system tab you have syslog and debug > server. > > Sent from my mobile device. > > On Jan 29, 2013, at 2:05 PM, Avi Marcus wrote: > > Any suggestions on how to figure this out? I'd really appreciate it! > > -Avi > > > On Mon, Jan 7, 2013 at 3:12 PM, Avi Marcus wrote: > >> Here's an intersting one that seems to have no results on google or in >> debug info: >> >> I have an SPA-2102 (firmware 5.2.10) that returns 486/Busy Here when a >> call comes in to the active line. >> >> "call waiting" is turned on, so it should be beeping in instead... >> >> I got these logs off the SPA but it doesn't seem very helpful: (and there >> should be more, I think) >> >> ==> /var/log/syslog <== >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 100 Trying >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 486 Busy Here >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >> >> >> I saw one suggestion to do a factory reset that fixed the problem, but >> that's a bit troublesome at the moment since the client can't >> load provisioning files (dumb network.) >> >> -Avi Marcus >> > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/1ea0e4b8/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue Jan 29 23:13:54 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 29 Jan 2013 20:13:54 +0000 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) Message-ID: Hello everyone, I've been recently reviewing several DID providers, and looking for some feedback. I appreciate this isn't totally on topic for FreeSWITCH, but over the last few months I've found the community feedback to be invaluable - so I'm hoping it might open up a good discussion. I would be really interested to hear other peoples experiences with their DID providers, as much detail as possible. There are several providers we've looked at, but they are either reselling others services, or do not deliver the service quality we are looking for. The criteria I've been looking for is; * Able to provide either US or UK numbers * Exposes an API that allows control of purchasing and routing * Guarantees premium call quality, good latency, and inbound CLI. * Does NOT operate on a market/exchange basis * Trading as a telecoms provider for more than 5 years * Has a reputable name within the community or industry So far, I have only found the following providers that meet this criteria; * Hello Telecom - UK numbers only, ITSPA registered, no charges for minutes, rental or concurrent channels. No minimum charges. Excellent call quality/latency, no CLI problems, and almost no downtime in the last 6 months. However, they do not offer a sane purchasing API which makes automation difficult, and interop with FS has been somewhat painful. * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but still very fair. Currently at inter-op stage, so can't comment further - but they seem very good so far. * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, difficult to justify unless you already have existing traffic. * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over 9000 pages of inter-op and around ?10k in set up fees. Thanks in advance Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/a2ccb05d/attachment.html From victor.chukalovskiy at gmail.com Tue Jan 29 23:21:44 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 29 Jan 2013 15:21:44 -0500 Subject: [Freeswitch-users] Rejecting undesired codecs in bypass media mode Message-ID: <51082F58.9020208@gmail.com> Hello, _Call scenario:_ call comes in with undesired codec only in SDP. (e.g. PCMA) In my sip profile I have: inbound-bypass-media="true" inbound-codec-negotiation="greedy" inbound-codec-prefs="PCMU,G729" _What happens:_ regardless of the codec in SDP, Freeswitch executes dialplan and sends the call to the B-leg. It also happens regardless of wether I use "late" or "early" negotiation _What I would like to see:_ FS rejecting the call right away if SDP does not match inbound-codec-prefs. Should not even hit the dial-plan. Any clue on how to achieve this? Many thanks, -Victor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/b857f449/attachment.html From avi at avimarcus.net Tue Jan 29 23:55:11 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 29 Jan 2013 22:55:11 +0200 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: Shouldn't go without mentioning ukddi.com... It's free, since it's subsidized by incoming traffic compensation. He claims that's more profitable than termination service is, easily allowing him to keep the business running. I've not seen the API, I think it has to be specially activated. He's very responsive to emails. -Avi On Tue, Jan 29, 2013 at 10:13 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello everyone, > > I've been recently reviewing several DID providers, and looking for some > feedback. > > I appreciate this isn't totally on topic for FreeSWITCH, but over the last > few months I've found the community feedback to be invaluable - so I'm > hoping it might open up a good discussion. > > I would be really interested to hear other peoples experiences with their > DID providers, as much detail as possible. > > There are several providers we've looked at, but they are either reselling > others services, or do not deliver the service quality we are looking for.. > > The criteria I've been looking for is; > > * Able to provide either US or UK numbers > * Exposes an API that allows control of purchasing and routing > * Guarantees premium call quality, good latency, and inbound CLI. > * Does NOT operate on a market/exchange basis > * Trading as a telecoms provider for more than 5 years > * Has a reputable name within the community or industry > > So far, I have only found the following providers that meet this criteria; > > * Hello Telecom - UK numbers only, ITSPA registered, no charges for > minutes, rental or concurrent channels. No minimum charges. Excellent call > quality/latency, no CLI problems, and almost no downtime in the last 6 > months. However, they do not offer a sane purchasing API which makes > automation difficult, and interop with FS has been somewhat painful. > > * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but > still very fair. Currently at inter-op stage, so can't comment further - > but they seem very good so far. > > * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, > difficult to justify unless you already have existing traffic. > > * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over > 9000 pages of inter-op and around ?10k in set up fees. > > Thanks in advance > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/76dfc529/attachment-0001.html From djbinter at gmail.com Wed Jan 30 00:00:33 2013 From: djbinter at gmail.com (DJB International) Date: Tue, 29 Jan 2013 13:00:33 -0800 Subject: [Freeswitch-users] Rejecting undesired codecs in bypass media mode In-Reply-To: <51082F58.9020208@gmail.com> References: <51082F58.9020208@gmail.com> Message-ID: and turn of inbound-bypass-media, then you can set the bypass_media in dialplan before you bridge. -djbinter On Tue, Jan 29, 2013 at 12:21 PM, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Hello, > > *Call scenario:* call comes in with undesired codec only in SDP. (e.g. > PCMA) > > In my sip profile I have: > inbound-bypass-media="true" > inbound-codec-negotiation="greedy" > inbound-codec-prefs="PCMU,G729" > > *What happens:* regardless of the codec in SDP, Freeswitch executes > dialplan and sends the call to the B-leg. > > It also happens regardless of wether I use "late" or "early" negotiation > > *What I would like to see:* > FS rejecting the call right away if SDP does not match > inbound-codec-prefs. Should not even hit the dial-plan. > > Any clue on how to achieve this? > > Many thanks, > -Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/ab3792a6/attachment.html From victor.chukalovskiy at gmail.com Wed Jan 30 00:09:11 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 29 Jan 2013 16:09:11 -0500 Subject: [Freeswitch-users] Rejecting undesired codecs in bypass media mode In-Reply-To: References: <51082F58.9020208@gmail.com> Message-ID: <51083A77.1070308@gmail.com> Ok, thanks! I thought about it too. This will likely work. But I prefer to keep inbound-bypass-media=true in SIP profile. Any way to do it this way? This is for simplicity and to avoid the need to set inbound-bypass-media in the dialplan. Thank you, -Victor On 13-01-29 04:00 PM, DJB International wrote: > and turn of > inbound-bypass-media, then you can set the bypass_media in dialplan > before you bridge. > > -djbinter > > > On Tue, Jan 29, 2013 at 12:21 PM, Victor Chukalovskiy > > > wrote: > > Hello, > > _Call scenario:_ call comes in with undesired codec only in SDP. > (e.g. PCMA) > > In my sip profile I have: > inbound-bypass-media="true" > inbound-codec-negotiation="greedy" > inbound-codec-prefs="PCMU,G729" > > _What happens:_ regardless of the codec in SDP, Freeswitch > executes dialplan and sends the call to the B-leg. > > It also happens regardless of wether I use "late" or "early" > negotiation > > _What I would like to see:_ > FS rejecting the call right away if SDP does not match > inbound-codec-prefs. Should not even hit the dial-plan. > > Any clue on how to achieve this? > > Many thanks, > -Victor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/a561e441/attachment.html From vallimamod.abdullah at imtelecom.fr Wed Jan 30 01:17:59 2013 From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH) Date: Tue, 29 Jan 2013 23:17:59 +0100 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: <8F409268-AD86-4F5F-AAF5-2C1FE8E7379E@imtelecom.fr> Hi, We are using voiptalk.org for inbound dids (UK, US, AU mostly.) They work with voxbone but without the minimum spend requirement. The only problem may be that they do not provide an API (but this need to be confirmed.) Best Regards, Vallimamod. On Jan 29, 2013, at 9:13 PM, Cal Leeming [Simplicity Media Ltd] wrote: > Hello everyone, > > I've been recently reviewing several DID providers, and looking for some feedback. > > I appreciate this isn't totally on topic for FreeSWITCH, but over the last few months I've found the community feedback to be invaluable - so I'm hoping it might open up a good discussion. > > I would be really interested to hear other peoples experiences with their DID providers, as much detail as possible. > > There are several providers we've looked at, but they are either reselling others services, or do not deliver the service quality we are looking for. > > The criteria I've been looking for is; > > * Able to provide either US or UK numbers > * Exposes an API that allows control of purchasing and routing > * Guarantees premium call quality, good latency, and inbound CLI. > * Does NOT operate on a market/exchange basis > * Trading as a telecoms provider for more than 5 years > * Has a reputable name within the community or industry > > So far, I have only found the following providers that meet this criteria; > > * Hello Telecom - UK numbers only, ITSPA registered, no charges for minutes, rental or concurrent channels. No minimum charges. Excellent call quality/latency, no CLI problems, and almost no downtime in the last 6 months. However, they do not offer a sane purchasing API which makes automation difficult, and interop with FS has been somewhat painful. > > * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but still very fair. Currently at inter-op stage, so can't comment further - but they seem very good so far. > > * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, difficult to justify unless you already have existing traffic. > > * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over 9000 pages of inter-op and around ?10k in set up fees. > > Thanks in advance > > Cal From luke.milbourne at gmail.com Wed Jan 30 00:36:10 2013 From: luke.milbourne at gmail.com (Luke Milbourne) Date: Tue, 29 Jan 2013 21:36:10 +0000 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) Message-ID: Hi Cal, Have you investigated AQL? I use them currently for my DID, not on the type of scale you are talking about but I have been impressed with their services so far. Luke On 29 January 2013 20:55, wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: SPA-2102 Gives 486 instead of Call Waiting (Gilad Abada) > 2. DID providers - any thoughts? (jan 2013) > (Cal Leeming [Simplicity Media Ltd]) > 3. Rejecting undesired codecs in bypass media mode > (Victor Chukalovskiy) > 4. Re: DID providers - any thoughts? (jan 2013) (Avi Marcus) > > > ---------- Forwarded message ---------- > From: Gilad Abada > To: FreeSWITCH Users Help > Cc: > Date: Tue, 29 Jan 2013 14:34:47 -0500 > Subject: Re: [Freeswitch-users] SPA-2102 Gives 486 instead of Call Waiting > I think if you turn debug and syslog on level 3 it gives more info > > Sent from my mobile device. > > On Jan 29, 2013, at 2:28 PM, Avi Marcus wrote: > > I did, that's how I got the log in my first email... > Was I supposed to find more than that? > -Avi > > > On Tue, Jan 29, 2013 at 9:08 PM, Gilad Abada wrote: > >> Avi, >> >> You can try to enable syslog trough the GUI and see what the spa 2102 os >> actually doing. I think it's under the system tab you have syslog and debug >> server. >> >> Sent from my mobile device. >> >> On Jan 29, 2013, at 2:05 PM, Avi Marcus wrote: >> >> Any suggestions on how to figure this out? I'd really appreciate it! >> >> -Avi >> >> >> On Mon, Jan 7, 2013 at 3:12 PM, Avi Marcus wrote: >> >>> Here's an intersting one that seems to have no results on google or in >>> debug info: >>> >>> I have an SPA-2102 (firmware 5.2.10) that returns 486/Busy Here when a >>> call comes in to the active line. >>> >>> "call waiting" is turned on, so it should be beeping in instead... >>> >>> I got these logs off the SPA but it doesn't seem very helpful: (and >>> there should be more, I think) >>> >>> ==> /var/log/syslog <== >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >>> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 100 Trying >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 486 Busy Here >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >>> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >>> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>> >>> >>> I saw one suggestion to do a factory reset that fixed the problem, but >>> that's a bit troublesome at the moment since the client can't >>> load provisioning files (dumb network.) >>> >>> -Avi Marcus >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ---------- Forwarded message ---------- > From: "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming at simplicitymedialtd.co.uk> > To: FreeSWITCH Users Help > Cc: > Date: Tue, 29 Jan 2013 20:13:54 +0000 > Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) > Hello everyone, > > I've been recently reviewing several DID providers, and looking for some > feedback. > > I appreciate this isn't totally on topic for FreeSWITCH, but over the last > few months I've found the community feedback to be invaluable - so I'm > hoping it might open up a good discussion. > > I would be really interested to hear other peoples experiences with their > DID providers, as much detail as possible. > > There are several providers we've looked at, but they are either reselling > others services, or do not deliver the service quality we are looking for. > > The criteria I've been looking for is; > > * Able to provide either US or UK numbers > * Exposes an API that allows control of purchasing and routing > * Guarantees premium call quality, good latency, and inbound CLI. > * Does NOT operate on a market/exchange basis > * Trading as a telecoms provider for more than 5 years > * Has a reputable name within the community or industry > > So far, I have only found the following providers that meet this criteria; > > * Hello Telecom - UK numbers only, ITSPA registered, no charges for > minutes, rental or concurrent channels. No minimum charges. Excellent call > quality/latency, no CLI problems, and almost no downtime in the last 6 > months. However, they do not offer a sane purchasing API which makes > automation difficult, and interop with FS has been somewhat painful. > > * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but > still very fair. Currently at inter-op stage, so can't comment further - > but they seem very good so far. > > * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, > difficult to justify unless you already have existing traffic. > > * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over > 9000 pages of inter-op and around ?10k in set up fees. > > Thanks in advance > > Cal > > > ---------- Forwarded message ---------- > From: Victor Chukalovskiy > To: FreeSWITCH Users Help > Cc: > Date: Tue, 29 Jan 2013 15:21:44 -0500 > Subject: [Freeswitch-users] Rejecting undesired codecs in bypass media mode > Hello, > > *Call scenario:* call comes in with undesired codec only in SDP. (e.g. > PCMA) > > In my sip profile I have: > inbound-bypass-media="true" > inbound-codec-negotiation="greedy" > inbound-codec-prefs="PCMU,G729" > > *What happens:* regardless of the codec in SDP, Freeswitch executes > dialplan and sends the call to the B-leg. > > It also happens regardless of wether I use "late" or "early" negotiation > > *What I would like to see:* > FS rejecting the call right away if SDP does not match > inbound-codec-prefs. Should not even hit the dial-plan. > > Any clue on how to achieve this? > > Many thanks, > -Victor > > > ---------- Forwarded message ---------- > From: Avi Marcus > To: FreeSWITCH Users Help > Cc: > Date: Tue, 29 Jan 2013 22:55:11 +0200 > Subject: Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013) > Shouldn't go without mentioning ukddi.com... > It's free, since it's subsidized by incoming traffic compensation. He > claims that's more profitable than termination service is, easily allowing > him to keep the business running. > I've not seen the API, I think it has to be specially activated. > > He's very responsive to emails. > > -Avi > > > On Tue, Jan 29, 2013 at 10:13 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello everyone, >> >> I've been recently reviewing several DID providers, and looking for some >> feedback. >> >> I appreciate this isn't totally on topic for FreeSWITCH, but over the >> last few months I've found the community feedback to be invaluable - so I'm >> hoping it might open up a good discussion. >> >> I would be really interested to hear other peoples experiences with their >> DID providers, as much detail as possible. >> >> There are several providers we've looked at, but they are either >> reselling others services, or do not deliver the service quality we are >> looking for. >> >> The criteria I've been looking for is; >> >> * Able to provide either US or UK numbers >> * Exposes an API that allows control of purchasing and routing >> * Guarantees premium call quality, good latency, and inbound CLI. >> * Does NOT operate on a market/exchange basis >> * Trading as a telecoms provider for more than 5 years >> * Has a reputable name within the community or industry >> >> So far, I have only found the following providers that meet this criteria; >> >> * Hello Telecom - UK numbers only, ITSPA registered, no charges for >> minutes, rental or concurrent channels. No minimum charges. Excellent call >> quality/latency, no CLI problems, and almost no downtime in the last 6 >> months. However, they do not offer a sane purchasing API which makes >> automation difficult, and interop with FS has been somewhat painful. >> >> * Simwood - UK numbers only, ITSPA registered. Slightly more expensive >> but still very fair. Currently at inter-op stage, so can't comment further >> - but they seem very good so far. >> >> * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, >> difficult to justify unless you already have existing traffic. >> >> * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over >> 9000 pages of inter-op and around ?10k in set up fees. >> >> Thanks in advance >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Luke Milbourne Email: luke.milbourne at gmail.com Tel: 07857154817 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/a561ed72/attachment-0001.html From anthony.minessale at gmail.com Wed Jan 30 02:14:22 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Jan 2013 17:14:22 -0600 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: Things like this are sad. We depend on testing and reporting for our releases. If you wait months to bring up a problem. It will spoil the whole release. If you you feel some leak has appeared suddenly, why can't you do git bisect and find it? On Tue, Jan 29, 2013 at 12:58 PM, Mario G wrote: > Probably not be related, but you never know: on OSX since Nov/Dec there > has been a memory leak on 1.2 and head that occurs for in/out/and > registrations. I had to triple memory and recycle FreeSwitch every 2-3 days > since then. Will open a Jira when I can obtain detailed info and possibly > run valgrind. I am also also waiting to update the main FS computer from > 10.6.8 to 10.8.3 to see its effect. Hopefully in Feb. > Mario G > > On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: > > Can you systematically increase the current call count and see where you > do see something? > On a system with limited ram you can also consider stripping all the .so > files in the mods and lib dir but you will need to put non-stripped ones in > for any debugging. > > I don't really see a correlation on how not using a timer could trigger an > sustained memory increase so that's why I'd like you to step up the number > and see if you can find a number of calls that tops out because usually > there always is a magic number where it will hover and go up and down a meg > at a time. > > SIP calls are required to keep state data around for at least 30 seconds > after a call ends and there are a number of pools in the code that inflate > once and do not return the memory. Its usually possible to identify the > high watermark on a particular box. > > For instance the machine we host the conference call on launches using 25 > megs and hovers at about 350 megs once it has accumulated all the pool > memory it needs over time. > > > > > > On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle wrote: > >> Version: >> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 >> (-n20130122T122521Z-1~squeeze+1) >> >> The calls are bridged, from one sofia profile to another. >> >> Unfortunately two concurrent calls doesn't seem to trigger the same >> behavior. >> >> >> - Jeff >> >> >> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: >> >>> Ok the package timestamps/versions there don?t do us a lot of good, we >>> need to know the version line from the FreeSWITCH CLI.. >>> >>> >>> On 1/28/13 9:59 PM, "Jeff Pyle" wrote: >>> >>> I just updated from repo.profhost.eu . The >>> most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same >>> behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. >>> After disconnecting the calls the usage returned to 8.5%. >>> >>> >>> I started toggling config items to see if I could impact this. I found >>> one that seems to have an effect: rtp-timer-name in the sofia profile >>> config. By changing it from 'soft' to 'none', the CPU utilization with 30 >>> calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. >>> >>> That's great, but does it make any sense? >>> >>> Does an rtp-timer-name of 'none' pose any risks? >>> >>> >>> >>> - Jeff >>> >>> >>> >>> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle >>> wrote: >>> >>> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >>> >>> >>> - Jeff >>> >>> >>> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner >>> wrote: >>> >>> Out of curiosity does your distro use uclibc, eglibc, or glibc? >>> >>> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle >>> wrote: >>> > Hello, >>> > >>> > I'm running HEAD version from Jan 22 on an Alix board with an AMD >>> Geode LX >>> > processor (i386). I can sustain 30 concurrent calls averaging around >>> 70% >>> > CPU utilization by the freeswitch process, measured by top. Bypass >>> media >>> > and proxy media are disabled. PCMU is forced on both endpoints (no >>> > transcoding). >>> > >>> > The problem is the RAM usage over time. The board has 256M. Idle, >>> > freeswitch occupies around 4% after a fresh restart. A minute or so >>> after >>> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, >>> 13.6%. >>> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM >>> usage >>> > to 6-8%. >>> > >>> > I've not tried to troubleshoot an issue like this before. Is valgrind >>> the >>> > next step, or would something else make more sense? >>> > >>> > >>> > Regards, >>> > Jeff >>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/d6182342/attachment.html From jpyle at fidelityvoice.com Wed Jan 30 02:15:06 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Tue, 29 Jan 2013 18:15:06 -0500 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: I gave you bad information before. It does happen with just one or two calls, it just takes a long time. Valgrind was not an option on the embedded box I was able to reproduce the issue on a regular x86_64 machine running squeeze, FreeSWITCH Version 1.3.4-n20130128T034121Z-1~squeeze+1 (-n20130128T034121Z-1~squeeze+1). I have the vg.log file produced with two calls up for approximately 35 minutes. No other SIP activity occurred during that time period. Is it Jira time, or is there other data you'd like to see? - Jeff On Tue, Jan 29, 2013 at 12:04 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Can you systematically increase the current call count and see where you > do see something? > On a system with limited ram you can also consider stripping all the .so > files in the mods and lib dir but you will need to put non-stripped ones in > for any debugging. > > I don't really see a correlation on how not using a timer could trigger an > sustained memory increase so that's why I'd like you to step up the number > and see if you can find a number of calls that tops out because usually > there always is a magic number where it will hover and go up and down a meg > at a time. > > SIP calls are required to keep state data around for at least 30 seconds > after a call ends and there are a number of pools in the code that inflate > once and do not return the memory. Its usually possible to identify the > high watermark on a particular box. > > For instance the machine we host the conference call on launches using 25 > megs and hovers at about 350 megs once it has accumulated all the pool > memory it needs over time. > > > > > > On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle wrote: > >> Version: >> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 >> (-n20130122T122521Z-1~squeeze+1) >> >> The calls are bridged, from one sofia profile to another. >> >> Unfortunately two concurrent calls doesn't seem to trigger the same >> behavior. >> >> >> - Jeff >> >> >> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: >> >>> Ok the package timestamps/versions there don?t do us a lot of good, we >>> need to know the version line from the FreeSWITCH CLI.. >>> >>> >>> On 1/28/13 9:59 PM, "Jeff Pyle" wrote: >>> >>> I just updated from repo.profhost.eu . The >>> most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same >>> behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. >>> After disconnecting the calls the usage returned to 8.5%. >>> >>> >>> I started toggling config items to see if I could impact this. I found >>> one that seems to have an effect: rtp-timer-name in the sofia profile >>> config. By changing it from 'soft' to 'none', the CPU utilization with 30 >>> calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. >>> >>> That's great, but does it make any sense? >>> >>> Does an rtp-timer-name of 'none' pose any risks? >>> >>> >>> >>> - Jeff >>> >>> >>> >>> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle >>> wrote: >>> >>> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >>> >>> >>> - Jeff >>> >>> >>> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner >>> wrote: >>> >>> Out of curiosity does your distro use uclibc, eglibc, or glibc? >>> >>> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle >>> wrote: >>> > Hello, >>> > >>> > I'm running HEAD version from Jan 22 on an Alix board with an AMD >>> Geode LX >>> > processor (i386). I can sustain 30 concurrent calls averaging around >>> 70% >>> > CPU utilization by the freeswitch process, measured by top. Bypass >>> media >>> > and proxy media are disabled. PCMU is forced on both endpoints (no >>> > transcoding). >>> > >>> > The problem is the RAM usage over time. The board has 256M. Idle, >>> > freeswitch occupies around 4% after a fresh restart. A minute or so >>> after >>> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, >>> 13.6%. >>> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM >>> usage >>> > to 6-8%. >>> > >>> > I've not tried to troubleshoot an issue like this before. Is valgrind >>> the >>> > next step, or would something else make more sense? >>> > >>> > >>> > Regards, >>> > Jeff >>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/3c5fb6cd/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Wed Jan 30 02:29:14 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 29 Jan 2013 23:29:14 +0000 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: Thank you everyone for the feedback, here is a few comments; @Avi - ukddi.com was down on the first visit, and I couldn't find much trading history behind the company.. The website now redirects to http://www.3c.co.uk but contains very little info. Duedil shows a 'Compulsory strike off' for 3c Limited, the address on the website shows different to the one on company records, and the registered address appears to be a residential. The site also contains no contact info, the whois shows a gmail address as the email contact, and the website is registered to an individual rather than a company (which is actually against Nominet rules!). Now interestingly, his domain points to 'telng.com' which is actually owned by Hello Telecom.. other people associated with TelNG are also connected with Hello Telecom. Either way, all these things set off a lot of alarm bells :/ @Vallimamod VoIPTalk (otherwise known as Telappliant) is for consumer/b2b, not really carrier grade trunking. Although they are re-selling Voxbone without the spend commitment, they are still just another reseller. @Luke Ah, now we had actually spoken with AQL on several occasions, but I was not overly impressed with their legal and sales teams (took weeks of emails to get clear/concise answers). Their concurrent channel charges were quite expensive, and overall we were left with a bad taste from it. However, that being said, they are certainly a reputable company and I couldn't find any negative reviews on them. We didn't get to the inter-op stage so I can't directly comment on service quality, but from what I could tell, their services seemed pretty solid. Cal On Tue, Jan 29, 2013 at 8:13 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello everyone, > > I've been recently reviewing several DID providers, and looking for some > feedback. > > I appreciate this isn't totally on topic for FreeSWITCH, but over the last > few months I've found the community feedback to be invaluable - so I'm > hoping it might open up a good discussion. > > I would be really interested to hear other peoples experiences with their > DID providers, as much detail as possible. > > There are several providers we've looked at, but they are either reselling > others services, or do not deliver the service quality we are looking for. > > The criteria I've been looking for is; > > * Able to provide either US or UK numbers > * Exposes an API that allows control of purchasing and routing > * Guarantees premium call quality, good latency, and inbound CLI. > * Does NOT operate on a market/exchange basis > * Trading as a telecoms provider for more than 5 years > * Has a reputable name within the community or industry > > So far, I have only found the following providers that meet this criteria; > > * Hello Telecom - UK numbers only, ITSPA registered, no charges for > minutes, rental or concurrent channels. No minimum charges. Excellent call > quality/latency, no CLI problems, and almost no downtime in the last 6 > months. However, they do not offer a sane purchasing API which makes > automation difficult, and interop with FS has been somewhat painful. > > * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but > still very fair. Currently at inter-op stage, so can't comment further - > but they seem very good so far. > > * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, > difficult to justify unless you already have existing traffic. > > * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over > 9000 pages of inter-op and around ?10k in set up fees. > > Thanks in advance > > Cal > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/1f4b7c13/attachment.html From anthony.minessale at gmail.com Wed Jan 30 02:32:07 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Jan 2013 17:32:07 -0600 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: Were are you getting the package from? 1.3 is on 13 right now so 4 is a bit too old. Can you build HEAD from source and reproduce a log you can post to jira? On Tue, Jan 29, 2013 at 5:15 PM, Jeff Pyle wrote: > I gave you bad information before. It does happen with just one or two > calls, it just takes a long time. Valgrind was not an option on the > embedded box I was able to reproduce the issue on a regular x86_64 machine > running squeeze, FreeSWITCH Version 1.3.4-n20130128T034121Z-1~squeeze+1 > (-n20130128T034121Z-1~squeeze+1). > > I have the vg.log file produced with two calls up for approximately 35 > minutes. No other SIP activity occurred during that time period. Is it > Jira time, or is there other data you'd like to see? > > > - Jeff > > > On Tue, Jan 29, 2013 at 12:04 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Can you systematically increase the current call count and see where you >> do see something? >> On a system with limited ram you can also consider stripping all the .so >> files in the mods and lib dir but you will need to put non-stripped ones in >> for any debugging. >> >> I don't really see a correlation on how not using a timer could trigger >> an sustained memory increase so that's why I'd like you to step up the >> number and see if you can find a number of calls that tops out because >> usually there always is a magic number where it will hover and go up and >> down a meg at a time. >> >> SIP calls are required to keep state data around for at least 30 seconds >> after a call ends and there are a number of pools in the code that inflate >> once and do not return the memory. Its usually possible to identify the >> high watermark on a particular box. >> >> For instance the machine we host the conference call on launches using 25 >> megs and hovers at about 350 megs once it has accumulated all the pool >> memory it needs over time. >> >> >> >> >> >> On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle wrote: >> >>> Version: >>> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 >>> (-n20130122T122521Z-1~squeeze+1) >>> >>> The calls are bridged, from one sofia profile to another. >>> >>> Unfortunately two concurrent calls doesn't seem to trigger the same >>> behavior. >>> >>> >>> - Jeff >>> >>> >>> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: >>> >>>> Ok the package timestamps/versions there don?t do us a lot of good, >>>> we need to know the version line from the FreeSWITCH CLI.. >>>> >>>> >>>> On 1/28/13 9:59 PM, "Jeff Pyle" wrote: >>>> >>>> I just updated from repo.profhost.eu . The >>>> most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same >>>> behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. >>>> After disconnecting the calls the usage returned to 8.5%. >>>> >>>> >>>> I started toggling config items to see if I could impact this. I found >>>> one that seems to have an effect: rtp-timer-name in the sofia profile >>>> config. By changing it from 'soft' to 'none', the CPU utilization with 30 >>>> calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. >>>> >>>> That's great, but does it make any sense? >>>> >>>> Does an rtp-timer-name of 'none' pose any risks? >>>> >>>> >>>> >>>> - Jeff >>>> >>>> >>>> >>>> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle >>>> wrote: >>>> >>>> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >>>> >>>> >>>> - Jeff >>>> >>>> >>>> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner >>>> wrote: >>>> >>>> Out of curiosity does your distro use uclibc, eglibc, or glibc? >>>> >>>> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle >>>> wrote: >>>> > Hello, >>>> > >>>> > I'm running HEAD version from Jan 22 on an Alix board with an AMD >>>> Geode LX >>>> > processor (i386). I can sustain 30 concurrent calls averaging around >>>> 70% >>>> > CPU utilization by the freeswitch process, measured by top. Bypass >>>> media >>>> > and proxy media are disabled. PCMU is forced on both endpoints (no >>>> > transcoding). >>>> > >>>> > The problem is the RAM usage over time. The board has 256M. Idle, >>>> > freeswitch occupies around 4% after a fresh restart. A minute or so >>>> after >>>> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, >>>> 13.6%. >>>> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM >>>> usage >>>> > to 6-8%. >>>> > >>>> > I've not tried to troubleshoot an issue like this before. Is >>>> valgrind the >>>> > next step, or would something else make more sense? >>>> > >>>> > >>>> > Regards, >>>> > Jeff >>>> >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/1b572aa8/attachment-0001.html From avi at avimarcus.net Wed Jan 30 02:40:01 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 30 Jan 2013 01:40:01 +0200 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: Re: UKDDI you can email Dave for more info: dave at 3c.co.uk -Avi On Wed, Jan 30, 2013 at 1:29 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Thank you everyone for the feedback, here is a few comments; > > @Avi - ukddi.com was down on the first visit, and I couldn't find much > trading history behind the company.. The website now redirects to > http://www.3c.co.uk but contains very little info. Duedil shows > a 'Compulsory strike off' for 3c Limited, the address on the website shows > different to the one on company records, and the registered address appears > to be a residential. The site also contains no contact info, the whois > shows a gmail address as the email contact, and the website is registered > to an individual rather than a company (which is actually against Nominet > rules!). Now interestingly, his domain points to 'telng.com' which is > actually owned by Hello Telecom.. other people associated with TelNG are > also connected with Hello Telecom. Either way, all these things set off a > lot of alarm bells :/ > > @Vallimamod VoIPTalk (otherwise known as Telappliant) is for consumer/b2b, > not really carrier grade trunking. Although they are re-selling Voxbone > without the spend commitment, they are still just another reseller. > > @Luke Ah, now we had actually spoken with AQL on several occasions, but I > was not overly impressed with their legal and sales teams (took weeks of > emails to get clear/concise answers). Their concurrent channel charges were > quite expensive, and overall we were left with a bad taste from it. > However, that being said, they are certainly a reputable company and I > couldn't find any negative reviews on them. We didn't get to the inter-op > stage so I can't directly comment on service quality, but from what I > could tell, their services seemed pretty solid. > > Cal > > > On Tue, Jan 29, 2013 at 8:13 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello everyone, >> >> I've been recently reviewing several DID providers, and looking for some >> feedback. >> >> I appreciate this isn't totally on topic for FreeSWITCH, but over the >> last few months I've found the community feedback to be invaluable - so I'm >> hoping it might open up a good discussion. >> >> I would be really interested to hear other peoples experiences with their >> DID providers, as much detail as possible. >> >> There are several providers we've looked at, but they are either >> reselling others services, or do not deliver the service quality we are >> looking for. >> >> The criteria I've been looking for is; >> >> * Able to provide either US or UK numbers >> * Exposes an API that allows control of purchasing and routing >> * Guarantees premium call quality, good latency, and inbound CLI. >> * Does NOT operate on a market/exchange basis >> * Trading as a telecoms provider for more than 5 years >> * Has a reputable name within the community or industry >> >> So far, I have only found the following providers that meet this criteria; >> >> * Hello Telecom - UK numbers only, ITSPA registered, no charges for >> minutes, rental or concurrent channels. No minimum charges. Excellent call >> quality/latency, no CLI problems, and almost no downtime in the last 6 >> months. However, they do not offer a sane purchasing API which makes >> automation difficult, and interop with FS has been somewhat painful. >> >> * Simwood - UK numbers only, ITSPA registered. Slightly more expensive >> but still very fair. Currently at inter-op stage, so can't comment further >> - but they seem very good so far. >> >> * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, >> difficult to justify unless you already have existing traffic. >> >> * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over >> 9000 pages of inter-op and around ?10k in set up fees. >> >> Thanks in advance >> >> Cal >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/179ad630/attachment.html From mthakershi at gmail.com Wed Jan 30 02:51:48 2013 From: mthakershi at gmail.com (Malay Thakershi) Date: Tue, 29 Jan 2013 17:51:48 -0600 Subject: [Freeswitch-users] Recording / concurrent calls and performance Message-ID: Hello, Does recording audio sessions slow down performance of concurrent calls / FreeSwitch? I am doing: application="record_session" in the dial plan before transferring the call to managed DLL. Thanks for help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/f6855710/attachment.html From mario_fs at mgtech.com Wed Jan 30 03:40:46 2013 From: mario_fs at mgtech.com (Mario G) Date: Tue, 29 Jan 2013 16:40:46 -0800 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: <2E3139C8-9CCD-4042-8FCD-F791BD4482B3@mgtech.com> Anthony, I tried bisect and the problem is that there are other issues (RTP error still in 1.2 stable, waiting for 1.2.6), and about 4-5 other issues in head that prevented it from working for a good amount of time. I started jiras for them and they are fixed, but doing bisect brings them back so it's had been really hard to pinpoint the memory issue. Believe me, I have put a LOT of time into trying to narrow it down before opening a JIRA and will continue to do so. Right now it's hard since there were personal emergencies the last 2 months so other pressing things had to take priority. Still, I am working on FreeSwitch keeping up-to-date with head in case other issues pop up I can open a JIRA on them. Mario G On Jan 29, 2013, at 3:14 PM, Anthony Minessale wrote: > Things like this are sad. We depend on testing and reporting for our releases. If you wait months to bring up a problem. It will spoil the whole release. > > If you you feel some leak has appeared suddenly, why can't you do git bisect and find it? > > > > On Tue, Jan 29, 2013 at 12:58 PM, Mario G wrote: > Probably not be related, but you never know: on OSX since Nov/Dec there has been a memory leak on 1.2 and head that occurs for in/out/and registrations. I had to triple memory and recycle FreeSwitch every 2-3 days since then. Will open a Jira when I can obtain detailed info and possibly run valgrind. I am also also waiting to update the main FS computer from 10.6.8 to 10.8.3 to see its effect. Hopefully in Feb. > Mario G > > On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: > >> Can you systematically increase the current call count and see where you do see something? >> On a system with limited ram you can also consider stripping all the .so files in the mods and lib dir but you will need to put non-stripped ones in for any debugging. >> >> I don't really see a correlation on how not using a timer could trigger an sustained memory increase so that's why I'd like you to step up the number and see if you can find a number of calls that tops out because usually there always is a magic number where it will hover and go up and down a meg at a time. >> >> SIP calls are required to keep state data around for at least 30 seconds after a call ends and there are a number of pools in the code that inflate once and do not return the memory. Its usually possible to identify the high watermark on a particular box. >> >> For instance the machine we host the conference call on launches using 25 megs and hovers at about 350 megs once it has accumulated all the pool memory it needs over time. >> >> >> >> >> >> On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle wrote: >> Version: >> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 (-n20130122T122521Z-1~squeeze+1) >> >> The calls are bridged, from one sofia profile to another. >> >> Unfortunately two concurrent calls doesn't seem to trigger the same behavior. >> >> >> - Jeff >> >> >> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: >> Ok the package timestamps/versions there don?t do us a lot of good, we need to know the version line from the FreeSWITCH CLI.. >> >> >> On 1/28/13 9:59 PM, "Jeff Pyle" wrote: >> >> I just updated from repo.profhost.eu . The most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. After disconnecting the calls the usage returned to 8.5%. >> >> >> I started toggling config items to see if I could impact this. I found one that seems to have an effect: rtp-timer-name in the sofia profile config. By changing it from 'soft' to 'none', the CPU utilization with 30 calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. >> >> That's great, but does it make any sense? >> >> Does an rtp-timer-name of 'none' pose any risks? >> >> >> >> - Jeff >> >> >> >> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle wrote: >> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >> >> >> - Jeff >> >> >> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner wrote: >> Out of curiosity does your distro use uclibc, eglibc, or glibc? >> >> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle wrote: >> > Hello, >> > >> > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode LX >> > processor (i386). I can sustain 30 concurrent calls averaging around 70% >> > CPU utilization by the freeswitch process, measured by top. Bypass media >> > and proxy media are disabled. PCMU is forced on both endpoints (no >> > transcoding). >> > >> > The problem is the RAM usage over time. The board has 256M. Idle, >> > freeswitch occupies around 4% after a fresh restart. A minute or so after >> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, 13.6%. >> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM usage >> > to 6-8%. >> > >> > I've not tried to troubleshoot an issue like this before. Is valgrind the >> > next step, or would something else make more sense? >> > >> > >> > Regards, >> > Jeff >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/0f3091f5/attachment-0001.html From shouldbeq931 at gmail.com Wed Jan 30 03:45:40 2013 From: shouldbeq931 at gmail.com (shouldbe q931) Date: Wed, 30 Jan 2013 00:45:40 +0000 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: On Tue, Jan 29, 2013 at 11:29 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Thank you everyone for the feedback, here is a few comments; > > @Avi - ukddi.com was down on the first visit, and I couldn't find much > trading history behind the company.. The website now redirects to > http://www.3c.co.uk but contains very little info. Duedil shows > a 'Compulsory strike off' for 3c Limited, the address on the website shows > different to the one on company records, and the registered address appears > to be a residential. The site also contains no contact info, the whois > shows a gmail address as the email contact, and the website is registered > to an individual rather than a company (which is actually against Nominet > rules!). Now interestingly, his domain points to 'telng.com' which is > actually owned by Hello Telecom.. other people associated with TelNG are > also connected with Hello Telecom. Either way, all these things set off a > lot of alarm bells :/ > > @Vallimamod VoIPTalk (otherwise known as Telappliant) is for consumer/b2b, > not really carrier grade trunking. Although they are re-selling Voxbone > without the spend commitment, they are still just another reseller. > > @Luke Ah, now we had actually spoken with AQL on several occasions, but I > was not overly impressed with their legal and sales teams (took weeks of > emails to get clear/concise answers). Their concurrent channel charges were > quite expensive, and overall we were left with a bad taste from it. > However, that being said, they are certainly a reputable company and I > couldn't find any negative reviews on them. We didn't get to the inter-op > stage so I can't directly comment on service quality, but from what I > could tell, their services seemed pretty solid. > > Cal > > > On Tue, Jan 29, 2013 at 8:13 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello everyone, >> >> I've been recently reviewing several DID providers, and looking for some >> feedback. >> >> I appreciate this isn't totally on topic for FreeSWITCH, but over the >> last few months I've found the community feedback to be invaluable - so I'm >> hoping it might open up a good discussion. >> >> I would be really interested to hear other peoples experiences with their >> DID providers, as much detail as possible. >> >> There are several providers we've looked at, but they are either >> reselling others services, or do not deliver the service quality we are >> looking for. >> >> The criteria I've been looking for is; >> >> * Able to provide either US or UK numbers >> * Exposes an API that allows control of purchasing and routing >> * Guarantees premium call quality, good latency, and inbound CLI. >> * Does NOT operate on a market/exchange basis >> * Trading as a telecoms provider for more than 5 years >> * Has a reputable name within the community or industry >> >> So far, I have only found the following providers that meet this criteria; >> >> * Hello Telecom - UK numbers only, ITSPA registered, no charges for >> minutes, rental or concurrent channels. No minimum charges. Excellent call >> quality/latency, no CLI problems, and almost no downtime in the last 6 >> months. However, they do not offer a sane purchasing API which makes >> automation difficult, and interop with FS has been somewhat painful. >> >> * Simwood - UK numbers only, ITSPA registered. Slightly more expensive >> but still very fair. Currently at inter-op stage, so can't comment further >> - but they seem very good so far. >> >> * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, >> difficult to justify unless you already have existing traffic. >> >> * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over >> 9000 pages of inter-op and around ?10k in set up fees. >> >> Thanks in advance >> >> Cal >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Small Gamma Telecom BT Equiv Virgin Media Business COLT I might also speak to Andrews and Arnold they might not quite fit your initial criteria, but well worth talking to www.aaisp.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/760ed353/attachment.html From freeswitch-list at puzzled.xs4all.nl Wed Jan 30 04:00:23 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 30 Jan 2013 02:00:23 +0100 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: <510870A7.3030607@puzzled.xs4all.nl> On 01/30/2013 12:29 AM, Cal Leeming [Simplicity Media Ltd] wrote: > Thank you everyone for the feedback, here is a few comments; Cal, How about http://www.magrathea-telecom.co.uk/ Not sure if they offer US numbers though. Regards, Patrick From ash at url.net.au Wed Jan 30 04:10:58 2013 From: ash at url.net.au (Ashley Breeden) Date: Wed, 30 Jan 2013 12:10:58 +1100 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: <510870A7.3030607@puzzled.xs4all.nl> References: <510870A7.3030607@puzzled.xs4all.nl> Message-ID: <4AA7184D-C10E-4ABC-8336-A9BE53357F6C@url.net.au> I find DIDww not too bad for ad-hoc DID's all over the world. Not sure if it meets your exact criteria. Check out www.didww.com Ash. On 30/01/2013, at 12:00 PM, Patrick Lists wrote: > On 01/30/2013 12:29 AM, Cal Leeming [Simplicity Media Ltd] wrote: >> Thank you everyone for the feedback, here is a few comments; > > Cal, > > How about http://www.magrathea-telecom.co.uk/ Not sure if they offer US > numbers though. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ga at steadfasttelecom.com Wed Jan 30 04:17:01 2013 From: ga at steadfasttelecom.com (Gilad Abada) Date: Tue, 29 Jan 2013 20:17:01 -0500 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: <510870A7.3030607@puzzled.xs4all.nl> References: <510870A7.3030607@puzzled.xs4all.nl> Message-ID: Hey You should check out flowroute.com also. Great price, great foot print, lots of DIDs including US and Canada. I think they have some 212 also! There support is also truly amazing. You can sign up for a free test account. I have been working with them for a few years now and I love them. On Tue, Jan 29, 2013 at 8:00 PM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 01/30/2013 12:29 AM, Cal Leeming [Simplicity Media Ltd] wrote: > > Thank you everyone for the feedback, here is a few comments; > > Cal, > > How about http://www.magrathea-telecom.co.uk/ Not sure if they offer US > numbers though. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 For over 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises. Make your virtual office a reality. Enjoy the freedom to travel while remaining connected to your office. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/b6907e03/attachment.html From krice at freeswitch.org Wed Jan 30 04:23:01 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 29 Jan 2013 19:23:01 -0600 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: <2E3139C8-9CCD-4042-8FCD-F791BD4482B3@mgtech.com> Message-ID: Mario, Is it fixed in the master branch? If it is fixed there it will be rolled down to 1.2 branch before long... On 1/29/13 6:40 PM, "Mario G" wrote: > Anthony, I tried bisect and the problem is that there are other issues (RTP > error still in 1.2 stable, waiting for 1.2.6), and about 4-5 other issues in > head that prevented it from working for a good amount of time. I started jiras > for them and they are fixed, but doing bisect brings them back so it's had > been really hard to pinpoint the memory issue. Believe me, I have put a LOT of > time into trying to narrow it down before opening a JIRA and will continue to > do so. Right now it's hard since there were personal emergencies the last 2 > months so other pressing things had to take priority. Still, I am working on > FreeSwitch keeping up-to-date with head in case other issues pop up I can open > a JIRA on them. > Mario G > > On Jan 29, 2013, at 3:14 PM, Anthony Minessale wrote: > >> Things like this are sad. We depend on testing and reporting for our >> releases. If you wait months to bring up a problem. It will spoil the whole >> release. >> >> If you you feel some leak has appeared suddenly, why can't you do git bisect >> and find it? >> >> >> >> On Tue, Jan 29, 2013 at 12:58 PM, Mario G wrote: >>> Probably not be related, but you never know: on OSX since Nov/Dec there has >>> been a memory leak on 1.2 and head that occurs for in/out/and registrations. >>> I had to triple memory and recycle FreeSwitch every 2-3 days since then. >>> Will open a Jira when I can obtain detailed info and possibly run valgrind. >>> I am also also waiting to update the main FS computer from 10.6.8 to 10.8.3 >>> to see its effect. Hopefully in Feb. >>> Mario G >>> >>> On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: >>> >>>> Can you systematically increase the current call count and see where you do >>>> see something? >>>> On a system with limited ram you can also consider stripping all the .so >>>> files in the mods and lib dir but you will need to put non-stripped ones in >>>> for any debugging. >>>> >>>> I don't really see a correlation on how not using a timer could trigger an >>>> sustained memory increase so that's why I'd like you to step up the number >>>> and see if you can find a number of calls that tops out because usually >>>> there always is a magic number where it will hover and go up and down a meg >>>> at a time. >>>> >>>> SIP calls are required to keep state data around for at least 30 seconds >>>> after a call ends and there are a number of pools in the code that inflate >>>> once and do not return the memory. Its usually possible to identify the >>>> high watermark on a particular box. >>>> >>>> For instance the machine we host the conference call on launches using 25 >>>> megs and hovers at about 350 megs once it has accumulated all the pool >>>> memory it needs over time. >>>> >>>> >>>> >>>> >>>> >>>> On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle wrote: >>>>> Version: >>>>> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 >>>>> (-n20130122T122521Z-1~squeeze+1) >>>>> >>>>> The calls are bridged, from one sofia profile to another. >>>>> >>>>> Unfortunately two concurrent calls doesn't seem to trigger the same >>>>> behavior. >>>>> >>>>> >>>>> - Jeff >>>>> >>>>> >>>>> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: >>>>>> Ok the package timestamps/versions there don?t do us a lot of good, we >>>>>> need to know the version line from the FreeSWITCH CLI.. >>>>>> >>>>>> >>>>>> On 1/28/13 9:59 PM, "Jeff Pyle" >>>>> > wrote: >>>>>> >>>>>>> I just updated from repo.profhost.eu >>>>>>> > . The most recent >>>>>>> timestamp on the packages was 2013-01-28 03:41:21 GMT. Same behavior. >>>>>>> At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. After >>>>>>> disconnecting the calls the usage returned to 8.5%. >>>>>>> >>>>>>> >>>>>>> I started toggling config items to see if I could impact this. I found >>>>>>> one that seems to have an effect: rtp-timer-name in the sofia profile >>>>>>> config. By changing it from 'soft' to 'none', the CPU utilization with >>>>>>> 30 calls dropped from ~70% to ~46%, and the RAM usage is rock solid at >>>>>>> 5.8%. >>>>>>> >>>>>>> That's great, but does it make any sense? >>>>>>> >>>>>>> Does an rtp-timer-name of 'none' pose any risks? >>>>>>> >>>>>>> >>>>>>> >>>>>>> - Jeff >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle >>>>>> > wrote: It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. - Jeff On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner > wrote: Out of curiosity does your distro use uclibc, eglibc, or glibc? On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle > wrote: > Hello, > > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode LX > processor (i386). I can sustain 30 concurrent calls averaging around 70% > CPU utilization by the freeswitch process, measured by top. Bypass media > and proxy media are disabled. PCMU is forced on both endpoints (no > transcoding). > > The problem is the RAM usage over time. The board has 256M. Idle, > freeswitch occupies around 4% after a fresh restart. A minute or so after > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, 13.6%. > After 60 minutes, near 65%. Disconnecting the calls returns the RAM usage > to 6-8%. > > I've not tried to troubleshoot an issue like this before. Is valgrind the > next step, or would something else make more sense? > > > Regards, > Jeff >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/2376c926/attachment-0001.html From anthony.minessale at gmail.com Wed Jan 30 04:41:02 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Jan 2013 19:41:02 -0600 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: <2E3139C8-9CCD-4042-8FCD-F791BD4482B3@mgtech.com> Message-ID: There is a commit today that found some sql leaks but I would not label them as massive. Its worth it to check and maybe run valgrind. From our perspective; Any time someone mentions nonchalantly, "Oh yeah I've had this glaring problem for months", breaks our heart =p I was just about to push the tag to unlock the 1.2.6 you so desperately want but now I am halting it to confirm this report. TEST TEST TEST HEAD... Today's head is tomorrow's stable.... On Tue, Jan 29, 2013 at 7:23 PM, Ken Rice wrote: > Mario, > > Is it fixed in the master branch? If it is fixed there it will be rolled > down to 1.2 branch before long... > > > On 1/29/13 6:40 PM, "Mario G" wrote: > > Anthony, I tried bisect and the problem is that there are other issues > (RTP error still in 1.2 stable, waiting for 1.2.6), and about 4-5 other > issues in head that prevented it from working for a good amount of time. I > started jiras for them and they are fixed, but doing bisect brings them > back so it's had been really hard to pinpoint the memory issue. Believe me, > I have put a LOT of time into trying to narrow it down before opening a > JIRA and will continue to do so. Right now it's hard since there were > personal emergencies the last 2 months so other pressing things had to take > priority. Still, I am working on FreeSwitch keeping up-to-date with head in > case other issues pop up I can open a JIRA on them. > Mario G > > On Jan 29, 2013, at 3:14 PM, Anthony Minessale wrote: > > Things like this are sad. We depend on testing and reporting for our > releases. If you wait months to bring up a problem. It will spoil the > whole release. > > If you you feel some leak has appeared suddenly, why can't you do git > bisect and find it? > > > > On Tue, Jan 29, 2013 at 12:58 PM, Mario G wrote: > > Probably not be related, but you never know: on OSX since Nov/Dec there > has been a memory leak on 1.2 and head that occurs for in/out/and > registrations. I had to triple memory and recycle FreeSwitch every 2-3 days > since then. Will open a Jira when I can obtain detailed info and possibly > run valgrind. I am also also waiting to update the main FS computer from > 10.6.8 to 10.8.3 to see its effect. Hopefully in Feb. > Mario G > > On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: > > Can you systematically increase the current call count and see where you > do see something? > On a system with limited ram you can also consider stripping all the .so > files in the mods and lib dir but you will need to put non-stripped ones in > for any debugging. > > I don't really see a correlation on how not using a timer could trigger an > sustained memory increase so that's why I'd like you to step up the number > and see if you can find a number of calls that tops out because usually > there always is a magic number where it will hover and go up and down a meg > at a time. > > SIP calls are required to keep state data around for at least 30 seconds > after a call ends and there are a number of pools in the code that inflate > once and do not return the memory. Its usually possible to identify the > high watermark on a particular box. > > For instance the machine we host the conference call on launches using 25 > megs and hovers at about 350 megs once it has accumulated all the pool > memory it needs over time. > > > > > > On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle > wrote: > > Version: > FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 > (-n20130122T122521Z-1~squeeze+1) > > The calls are bridged, from one sofia profile to another. > > Unfortunately two concurrent calls doesn't seem to trigger the same > behavior. > > > - Jeff > > > On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: > > Ok the package timestamps/versions there don?t do us a lot of good, we > need to know the version line from the FreeSWITCH CLI.. > > > On 1/28/13 9:59 PM, "Jeff Pyle" http://jpyle at fidelityvoice.com/> > wrote: > > I just updated from repo.profhost.eu < > http://repo.profhost.eu > . The most recent > timestamp on the packages was 2013-01-28 03:41:21 GMT. Same behavior. At > 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. After > disconnecting the calls the usage returned to 8.5%. > > > > I started toggling config items to see if I could impact this. I found > one that seems to have an effect: rtp-timer-name in the sofia profile > config. By changing it from 'soft' to 'none', the CPU utilization with 30 > calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. > > That's great, but does it make any sense? > > Does an rtp-timer-name of 'none' pose any risks? > > > > - Jeff > > > > On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle http://jpyle at fidelityvoice.com/> > wrote: > > It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. > > > - Jeff > > > On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner http://kris at kriskinc.com/> > wrote: > Out of curiosity does your distro use uclibc, eglibc, or glibc? > > On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle http://jpyle at fidelityvoice.com/> > wrote: > > Hello, > > > > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode > LX > > processor (i386). I can sustain 30 concurrent calls averaging around 70% > > CPU utilization by the freeswitch process, measured by top. Bypass media > > and proxy media are disabled. PCMU is forced on both endpoints (no > > transcoding). > > > > The problem is the RAM usage over time. The board has 256M. Idle, > > freeswitch occupies around 4% after a fresh restart. A minute or so > after > > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, > 13.6%. > > After 60 minutes, near 65%. Disconnecting the calls returns the RAM > usage > > to 6-8%. > > > > I've not tried to troubleshoot an issue like this before. Is valgrind > the > > next step, or would something else make more sense? > > > > > > Regards, > > Jeff > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/e3ae075d/attachment.html From mario_fs at mgtech.com Wed Jan 30 04:45:13 2013 From: mario_fs at mgtech.com (Mario G) Date: Tue, 29 Jan 2013 17:45:13 -0800 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: <2E21B6E0-251F-4200-B23D-D924127A1E9F@mgtech.com> Ken, I assume you mean the RTP error (FS 4900 4921, 4907), yes it has been fixed for a while. I thought I read somewhere it will go into 1.2.6. Sorry to go off-topic on this thread. Mario G On Jan 29, 2013, at 5:23 PM, Ken Rice wrote: > Mario, > > Is it fixed in the master branch? If it is fixed there it will be rolled down to 1.2 branch before long... > > > On 1/29/13 6:40 PM, "Mario G" wrote: > >> Anthony, I tried bisect and the problem is that there are other issues (RTP error still in 1.2 stable, waiting for 1.2.6), and about 4-5 other issues in head that prevented it from working for a good amount of time. I started jiras for them and they are fixed, but doing bisect brings them back so it's had been really hard to pinpoint the memory issue. Believe me, I have put a LOT of time into trying to narrow it down before opening a JIRA and will continue to do so. Right now it's hard since there were personal emergencies the last 2 months so other pressing things had to take priority. Still, I am working on FreeSwitch keeping up-to-date with head in case other issues pop up I can open a JIRA on them. >> Mario G >> >> On Jan 29, 2013, at 3:14 PM, Anthony Minessale wrote: >> >>> Things like this are sad. We depend on testing and reporting for our releases. If you wait months to bring up a problem. It will spoil the whole release. >>> >>> If you you feel some leak has appeared suddenly, why can't you do git bisect and find it? >>> >>> >>> >>> On Tue, Jan 29, 2013 at 12:58 PM, Mario G wrote: >>>> Probably not be related, but you never know: on OSX since Nov/Dec there has been a memory leak on 1.2 and head that occurs for in/out/and registrations. I had to triple memory and recycle FreeSwitch every 2-3 days since then. Will open a Jira when I can obtain detailed info and possibly run valgrind. I am also also waiting to update the main FS computer from 10.6.8 to 10.8.3 to see its effect. Hopefully in Feb. >>>> Mario G >>>> >>>> On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: >>>> >>>>> Can you systematically increase the current call count and see where you do see something? >>>>> On a system with limited ram you can also consider stripping all the .so files in the mods and lib dir but you will need to put non-stripped ones in for any debugging. >>>>> >>>>> I don't really see a correlation on how not using a timer could trigger an sustained memory increase so that's why I'd like you to step up the number and see if you can find a number of calls that tops out because usually there always is a magic number where it will hover and go up and down a meg at a time. >>>>> >>>>> SIP calls are required to keep state data around for at least 30 seconds after a call ends and there are a number of pools in the code that inflate once and do not return the memory. Its usually possible to identify the high watermark on a particular box. >>>>> >>>>> For instance the machine we host the conference call on launches using 25 megs and hovers at about 350 megs once it has accumulated all the pool memory it needs over time. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle wrote: >>>>>> Version: >>>>>> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 (-n20130122T122521Z-1~squeeze+1) >>>>>> >>>>>> The calls are bridged, from one sofia profile to another. >>>>>> >>>>>> Unfortunately two concurrent calls doesn't seem to trigger the same behavior. >>>>>> >>>>>> >>>>>> - Jeff >>>>>> >>>>>> >>>>>> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: >>>>>>> Ok the package timestamps/versions there don?t do us a lot of good, we need to know the version line from the FreeSWITCH CLI.. >>>>>>> >>>>>>> >>>>>>> On 1/28/13 9:59 PM, "Jeff Pyle" > wrote: >>>>>>> >>>>>>>> I just updated from repo.profhost.eu > . The most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. After disconnecting the calls the usage returned to 8.5%. >>>>>>>> >>>>>>>> >>>>>>>> I started toggling config items to see if I could impact this. I found one that seems to have an effect: rtp-timer-name in the sofia profile config. By changing it from 'soft' to 'none', the CPU utilization with 30 calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. >>>>>>>> >>>>>>>> That's great, but does it make any sense? >>>>>>>> >>>>>>>> Does an rtp-timer-name of 'none' pose any risks? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> - Jeff >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle > wrote: > It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. > > > - Jeff > > > On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner > wrote: > Out of curiosity does your distro use uclibc, eglibc, or glibc? > > On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle > wrote: > > Hello, > > > > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode LX > > processor (i386). I can sustain 30 concurrent calls averaging around 70% > > CPU utilization by the freeswitch process, measured by top. Bypass media > > and proxy media are disabled. PCMU is forced on both endpoints (no > > transcoding). > > > > The problem is the RAM usage over time. The board has 256M. Idle, > > freeswitch occupies around 4% after a fresh restart. A minute or so after > > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, 13.6%. > > After 60 minutes, near 65%. Disconnecting the calls returns the RAM usage > > to 6-8%. > > > > I've not tried to troubleshoot an issue like this before. Is valgrind the > > next step, or would something else make more sense? > > > > > > Regards, > > Jeff >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/6923ed98/attachment-0001.html From mario_fs at mgtech.com Wed Jan 30 05:09:49 2013 From: mario_fs at mgtech.com (Mario G) Date: Tue, 29 Jan 2013 18:09:49 -0800 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: <2E3139C8-9CCD-4042-8FCD-F791BD4482B3@mgtech.com> Message-ID: <06CDE1C3-916A-45B6-927E-9F59449305A1@mgtech.com> I have updated and if by some miracle it's fixed I will report back. Storage increases with no calls so I assume it's registrations (internal/external?). Please understand this is probably this worse week of the year for me to work on FS (or anything else). I promise to try and narrow down the commit that started it. The memory issue started at a bad time when I could not work on FS (emergencies) and by the time I updated other errors crept in that prevented use and I had to open JIRAs for them, so testing memory was out. Thanks for letting me know about todays fix. Mario G On Jan 29, 2013, at 5:41 PM, Anthony Minessale wrote: > There is a commit today that found some sql leaks but I would not label them as massive. Its worth it to check and maybe run valgrind. From our perspective; Any time someone mentions nonchalantly, "Oh yeah I've had this glaring problem for months", breaks our heart =p > I was just about to push the tag to unlock the 1.2.6 you so desperately want but now I am halting it to confirm this report. > > TEST TEST TEST HEAD... Today's head is tomorrow's stable.... > > > > On Tue, Jan 29, 2013 at 7:23 PM, Ken Rice wrote: > Mario, > > Is it fixed in the master branch? If it is fixed there it will be rolled down to 1.2 branch before long... > > > On 1/29/13 6:40 PM, "Mario G" wrote: > > Anthony, I tried bisect and the problem is that there are other issues (RTP error still in 1.2 stable, waiting for 1.2.6), and about 4-5 other issues in head that prevented it from working for a good amount of time. I started jiras for them and they are fixed, but doing bisect brings them back so it's had been really hard to pinpoint the memory issue. Believe me, I have put a LOT of time into trying to narrow it down before opening a JIRA and will continue to do so. Right now it's hard since there were personal emergencies the last 2 months so other pressing things had to take priority. Still, I am working on FreeSwitch keeping up-to-date with head in case other issues pop up I can open a JIRA on them. > Mario G > > On Jan 29, 2013, at 3:14 PM, Anthony Minessale wrote: > > Things like this are sad. We depend on testing and reporting for our releases. If you wait months to bring up a problem. It will spoil the whole release. > > If you you feel some leak has appeared suddenly, why can't you do git bisect and find it? > > > > On Tue, Jan 29, 2013 at 12:58 PM, Mario G wrote: > Probably not be related, but you never know: on OSX since Nov/Dec there has been a memory leak on 1.2 and head that occurs for in/out/and registrations. I had to triple memory and recycle FreeSwitch every 2-3 days since then. Will open a Jira when I can obtain detailed info and possibly run valgrind. I am also also waiting to update the main FS computer from 10.6.8 to 10.8.3 to see its effect. Hopefully in Feb. > Mario G > > On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: > > Can you systematically increase the current call count and see where you do see something? > On a system with limited ram you can also consider stripping all the .so files in the mods and lib dir but you will need to put non-stripped ones in for any debugging. > > I don't really see a correlation on how not using a timer could trigger an sustained memory increase so that's why I'd like you to step up the number and see if you can find a number of calls that tops out because usually there always is a magic number where it will hover and go up and down a meg at a time. > > SIP calls are required to keep state data around for at least 30 seconds after a call ends and there are a number of pools in the code that inflate once and do not return the memory. Its usually possible to identify the high watermark on a particular box. > > For instance the machine we host the conference call on launches using 25 megs and hovers at about 350 megs once it has accumulated all the pool memory it needs over time. > > > > > > On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle wrote: > Version: > FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 (-n20130122T122521Z-1~squeeze+1) > > The calls are bridged, from one sofia profile to another. > > Unfortunately two concurrent calls doesn't seem to trigger the same behavior. > > > - Jeff > > > On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: > Ok the package timestamps/versions there don?t do us a lot of good, we need to know the version line from the FreeSWITCH CLI.. > > > On 1/28/13 9:59 PM, "Jeff Pyle" > wrote: > > I just updated from repo.profhost.eu > . The most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. After disconnecting the calls the usage returned to 8.5%. > > > > I started toggling config items to see if I could impact this. I found one that seems to have an effect: rtp-timer-name in the sofia profile config. By changing it from 'soft' to 'none', the CPU utilization with 30 calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. > > That's great, but does it make any sense? > > Does an rtp-timer-name of 'none' pose any risks? > > > > - Jeff > > > > On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle > wrote: > It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. > > > - Jeff > > > On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner > wrote: > Out of curiosity does your distro use uclibc, eglibc, or glibc? > > On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle > wrote: > > Hello, > > > > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode LX > > processor (i386). I can sustain 30 concurrent calls averaging around 70% > > CPU utilization by the freeswitch process, measured by top. Bypass media > > and proxy media are disabled. PCMU is forced on both endpoints (no > > transcoding). > > > > The problem is the RAM usage over time. The board has 256M. Idle, > > freeswitch occupies around 4% after a fresh restart. A minute or so after > > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, 13.6%. > > After 60 minutes, near 65%. Disconnecting the calls returns the RAM usage > > to 6-8%. > > > > I've not tried to troubleshoot an issue like this before. Is valgrind the > > next step, or would something else make more sense? > > > > > > Regards, > > Jeff > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/f66be240/attachment.html From anthony.minessale at gmail.com Wed Jan 30 05:15:10 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 Jan 2013 20:15:10 -0600 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: <06CDE1C3-916A-45B6-927E-9F59449305A1@mgtech.com> References: <2E3139C8-9CCD-4042-8FCD-F791BD4482B3@mgtech.com> <06CDE1C3-916A-45B6-927E-9F59449305A1@mgtech.com> Message-ID: If its still there on head, try valgrind. valgrind --tool=memcheck --log-file=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes /usr/local/freeswitch/bin/freeswitch -vg On Tue, Jan 29, 2013 at 8:09 PM, Mario G wrote: > I have updated and if by some miracle it's fixed I will report back. > Storage increases with no calls so I assume it's registrations > (internal/external?). Please understand this is probably this worse week of > the year for me to work on FS (or anything else). I promise to try and > narrow down the commit that started it. The memory issue started at a bad > time when I could not work on FS (emergencies) and by the time I updated > other errors crept in that prevented use and I had to open JIRAs for them, > so testing memory was out. Thanks for letting me know about todays fix. > Mario G > > > On Jan 29, 2013, at 5:41 PM, Anthony Minessale wrote: > > There is a commit today that found some sql leaks but I would not label > them as massive. Its worth it to check and maybe run valgrind. From our > perspective; Any time someone mentions nonchalantly, "Oh yeah I've had this > glaring problem for months", breaks our heart =p > I was just about to push the tag to unlock the 1.2.6 you so desperately > want but now I am halting it to confirm this report. > > TEST TEST TEST HEAD... Today's head is tomorrow's stable.... > > > > On Tue, Jan 29, 2013 at 7:23 PM, Ken Rice wrote: > >> Mario, >> >> Is it fixed in the master branch? If it is fixed there it will be rolled >> down to 1.2 branch before long... >> >> >> On 1/29/13 6:40 PM, "Mario G" wrote: >> >> Anthony, I tried bisect and the problem is that there are other issues >> (RTP error still in 1.2 stable, waiting for 1.2.6), and about 4-5 other >> issues in head that prevented it from working for a good amount of time. I >> started jiras for them and they are fixed, but doing bisect brings them >> back so it's had been really hard to pinpoint the memory issue. Believe me, >> I have put a LOT of time into trying to narrow it down before opening a >> JIRA and will continue to do so. Right now it's hard since there were >> personal emergencies the last 2 months so other pressing things had to take >> priority. Still, I am working on FreeSwitch keeping up-to-date with head in >> case other issues pop up I can open a JIRA on them. >> Mario G >> >> On Jan 29, 2013, at 3:14 PM, Anthony Minessale wrote: >> >> Things like this are sad. We depend on testing and reporting for our >> releases. If you wait months to bring up a problem. It will spoil the >> whole release. >> >> If you you feel some leak has appeared suddenly, why can't you do git >> bisect and find it? >> >> >> >> On Tue, Jan 29, 2013 at 12:58 PM, Mario G wrote: >> >> Probably not be related, but you never know: on OSX since Nov/Dec there >> has been a memory leak on 1.2 and head that occurs for in/out/and >> registrations. I had to triple memory and recycle FreeSwitch every 2-3 days >> since then. Will open a Jira when I can obtain detailed info and possibly >> run valgrind. I am also also waiting to update the main FS computer from >> 10.6.8 to 10.8.3 to see its effect. Hopefully in Feb. >> Mario G >> >> On Jan 29, 2013, at 9:04 AM, Anthony Minessale wrote: >> >> Can you systematically increase the current call count and see where you >> do see something? >> On a system with limited ram you can also consider stripping all the .so >> files in the mods and lib dir but you will need to put non-stripped ones in >> for any debugging. >> >> I don't really see a correlation on how not using a timer could trigger >> an sustained memory increase so that's why I'd like you to step up the >> number and see if you can find a number of calls that tops out because >> usually there always is a magic number where it will hover and go up and >> down a meg at a time. >> >> SIP calls are required to keep state data around for at least 30 seconds >> after a call ends and there are a number of pools in the code that inflate >> once and do not return the memory. Its usually possible to identify the >> high watermark on a particular box. >> >> For instance the machine we host the conference call on launches using 25 >> megs and hovers at about 350 megs once it has accumulated all the pool >> memory it needs over time. >> >> >> >> >> >> On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle >> wrote: >> >> Version: >> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 >> (-n20130122T122521Z-1~squeeze+1) >> >> The calls are bridged, from one sofia profile to another. >> >> Unfortunately two concurrent calls doesn't seem to trigger the same >> behavior. >> >> >> - Jeff >> >> >> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: >> >> Ok the package timestamps/versions there don?t do us a lot of good, we >> need to know the version line from the FreeSWITCH CLI.. >> >> >> On 1/28/13 9:59 PM, "Jeff Pyle" > http://jpyle at fidelityvoice.com/> > wrote: >> >> I just updated from repo.profhost.eu < >> http://repo.profhost.eu > . The most recent >> timestamp on the packages was 2013-01-28 03:41:21 GMT. Same behavior. At >> 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. After >> disconnecting the calls the usage returned to 8.5%. >> >> >> >> I started toggling config items to see if I could impact this. I found >> one that seems to have an effect: rtp-timer-name in the sofia profile >> config. By changing it from 'soft' to 'none', the CPU utilization with 30 >> calls dropped from ~70% to ~46%, and the RAM usage is rock solid at 5.8%. >> >> That's great, but does it make any sense? >> >> Does an rtp-timer-name of 'none' pose any risks? >> >> >> >> - Jeff >> >> >> >> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle > http://jpyle at fidelityvoice.com/> > wrote: >> >> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >> >> >> - Jeff >> >> >> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner > http://kris at kriskinc.com/> > wrote: >> Out of curiosity does your distro use uclibc, eglibc, or glibc? >> >> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle > http://jpyle at fidelityvoice.com/> > wrote: >> > Hello, >> > >> > I'm running HEAD version from Jan 22 on an Alix board with an AMD Geode >> LX >> > processor (i386). I can sustain 30 concurrent calls averaging around >> 70% >> > CPU utilization by the freeswitch process, measured by top. Bypass >> media >> > and proxy media are disabled. PCMU is forced on both endpoints (no >> > transcoding). >> > >> > The problem is the RAM usage over time. The board has 256M. Idle, >> > freeswitch occupies around 4% after a fresh restart. A minute or so >> after >> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 minutes, >> 13.6%. >> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM >> usage >> > to 6-8%. >> > >> > I've not tried to troubleshoot an issue like this before. Is valgrind >> the >> > next step, or would something else make more sense? >> > >> > >> > Regards, >> > Jeff >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/5a1f429d/attachment-0001.html From jpyle at fidelityvoice.com Wed Jan 30 06:21:23 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Tue, 29 Jan 2013 22:21:23 -0500 Subject: [Freeswitch-users] unexplained RAM usage increase In-Reply-To: References: Message-ID: I installed via apt-get from http://repo.profhost.eu/debian. The same issue occurs on a fresh download from HEAD, yielding v1.3.13b. Jira FS-5065 . Log uploads to follow once some calls have had enough time to generate some data. - Jeff On Tue, Jan 29, 2013 at 6:32 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Were are you getting the package from? 1.3 is on 13 right now so 4 is a > bit too old. > Can you build HEAD from source and reproduce a log you can post to jira? > > > On Tue, Jan 29, 2013 at 5:15 PM, Jeff Pyle wrote: > >> I gave you bad information before. It does happen with just one or two >> calls, it just takes a long time. Valgrind was not an option on the >> embedded box I was able to reproduce the issue on a regular x86_64 machine >> running squeeze, FreeSWITCH Version 1.3.4-n20130128T034121Z-1~squeeze+1 >> (-n20130128T034121Z-1~squeeze+1). >> >> I have the vg.log file produced with two calls up for approximately 35 >> minutes. No other SIP activity occurred during that time period. Is it >> Jira time, or is there other data you'd like to see? >> >> >> - Jeff >> >> >> On Tue, Jan 29, 2013 at 12:04 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Can you systematically increase the current call count and see where you >>> do see something? >>> On a system with limited ram you can also consider stripping all the .so >>> files in the mods and lib dir but you will need to put non-stripped ones in >>> for any debugging. >>> >>> I don't really see a correlation on how not using a timer could trigger >>> an sustained memory increase so that's why I'd like you to step up the >>> number and see if you can find a number of calls that tops out because >>> usually there always is a magic number where it will hover and go up and >>> down a meg at a time. >>> >>> SIP calls are required to keep state data around for at least 30 seconds >>> after a call ends and there are a number of pools in the code that inflate >>> once and do not return the memory. Its usually possible to identify the >>> high watermark on a particular box. >>> >>> For instance the machine we host the conference call on launches using >>> 25 megs and hovers at about 350 megs once it has accumulated all the pool >>> memory it needs over time. >>> >>> >>> >>> >>> >>> On Tue, Jan 29, 2013 at 8:33 AM, Jeff Pyle wrote: >>> >>>> Version: >>>> FreeSWITCH version: 1.3.4-n20130122T122521Z-1~squeeze+1 >>>> (-n20130122T122521Z-1~squeeze+1) >>>> >>>> The calls are bridged, from one sofia profile to another. >>>> >>>> Unfortunately two concurrent calls doesn't seem to trigger the same >>>> behavior. >>>> >>>> >>>> - Jeff >>>> >>>> >>>> On Mon, Jan 28, 2013 at 11:21 PM, Ken Rice wrote: >>>> >>>>> Ok the package timestamps/versions there don?t do us a lot of good, >>>>> we need to know the version line from the FreeSWITCH CLI.. >>>>> >>>>> >>>>> On 1/28/13 9:59 PM, "Jeff Pyle" wrote: >>>>> >>>>> I just updated from repo.profhost.eu . The >>>>> most recent timestamp on the packages was 2013-01-28 03:41:21 GMT. Same >>>>> behavior. At 5 minutes it was using 12.5% RAM. At 40 minutes, 60.4%. >>>>> After disconnecting the calls the usage returned to 8.5%. >>>>> >>>>> >>>>> I started toggling config items to see if I could impact this. I >>>>> found one that seems to have an effect: rtp-timer-name in the sofia >>>>> profile config. By changing it from 'soft' to 'none', the CPU utilization >>>>> with 30 calls dropped from ~70% to ~46%, and the RAM usage is rock solid at >>>>> 5.8%. >>>>> >>>>> That's great, but does it make any sense? >>>>> >>>>> Does an rtp-timer-name of 'none' pose any risks? >>>>> >>>>> >>>>> >>>>> - Jeff >>>>> >>>>> >>>>> >>>>> On Mon, Jan 28, 2013 at 8:43 PM, Jeff Pyle >>>>> wrote: >>>>> >>>>> It's on Voyage Linux, a cousin of Debian. I believe it uses glibc. >>>>> >>>>> >>>>> - Jeff >>>>> >>>>> >>>>> On Mon, Jan 28, 2013 at 6:45 PM, Kristian Kielhofner < >>>>> kris at kriskinc.com> wrote: >>>>> >>>>> Out of curiosity does your distro use uclibc, eglibc, or glibc? >>>>> >>>>> On Mon, Jan 28, 2013 at 5:41 PM, Jeff Pyle >>>>> wrote: >>>>> > Hello, >>>>> > >>>>> > I'm running HEAD version from Jan 22 on an Alix board with an AMD >>>>> Geode LX >>>>> > processor (i386). I can sustain 30 concurrent calls averaging >>>>> around 70% >>>>> > CPU utilization by the freeswitch process, measured by top. Bypass >>>>> media >>>>> > and proxy media are disabled. PCMU is forced on both endpoints (no >>>>> > transcoding). >>>>> > >>>>> > The problem is the RAM usage over time. The board has 256M. Idle, >>>>> > freeswitch occupies around 4% after a fresh restart. A minute or so >>>>> after >>>>> > 30 calls are nailed up the RAM usage is about 7.2%. After 5 >>>>> minutes, 13.6%. >>>>> > After 60 minutes, near 65%. Disconnecting the calls returns the RAM >>>>> usage >>>>> > to 6-8%. >>>>> > >>>>> > I've not tried to troubleshoot an issue like this before. Is >>>>> valgrind the >>>>> > next step, or would something else make more sense? >>>>> > >>>>> > >>>>> > Regards, >>>>> > Jeff >>>>> >>>>> >>>>> >>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130129/23edd121/attachment.html From Sirish.MasurMohan at oa.com.au Wed Jan 30 09:23:52 2013 From: Sirish.MasurMohan at oa.com.au (Sirish Masur Mohan) Date: Wed, 30 Jan 2013 17:23:52 +1100 Subject: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error References: <965759A53E43FE439E43565A7715E5F0591C085DB2@oa-exchange1.oa.com.au> Message-ID: <965759A53E43FE439E43565A7715E5F0591C085EBB@oa-exchange1.oa.com.au> Ok, I have been able to temporarily work around this problem - in /usr/local/freeswitch/conf/autoload_configs/event_socket.conf.xml, when I change the listen-port from 8021 to any other port(with listen-IP being 0.0.0.0), SNOM calls go through smoothly - if they are set to 8021, I get a BEARERCAPABILITY_NOTAUTH problem. However, I need port 8021 as external applications are dependant on the same.. Would appreciate any inputs. With regards, Sirish From: Sirish Masur Mohan Sent: Tuesday, 29 January 2013 5:27 PM To: 'FreeSWITCH Users Help' Subject: RE: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error Hi Michael, I had a 'sleep' action in my dialplan, which seemed to cause the BEARERCAPABILITY_NOTAUTH error. If I comment it out, the error goes away. However, it makes way to a new error, '[CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]' My dial plan is as follows, am I missing something? With regards, Sirish From: Sirish Masur Mohan Sent: Tuesday, 29 January 2013 2:47 PM To: 'FreeSWITCH Users Help' Subject: RE: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error Hi Michael, As an experiment, I tried creating specific extensions for each phone, and then having specific dial plan, such as, With these changes, when I dial from one SNOM phone to other, the destination phone rings for a short time (say a second), after which the source phone shows a 403 forbidden error. Paste-bin of this new behaviour: http://pastebin.freeswitch.org/20515 With regards, Sirish From: Sirish Masur Mohan Sent: Tuesday, 29 January 2013 9:46 AM To: FreeSWITCH Users Help Subject: RE: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error Hi Michael, I have put the logs and SIP trace in the pastebin: http://pastebin.freeswitch.org/20514 I haven't created extensions for the SNOM phones in freeswitch, as the ACL for registration and incoming are Open - could this be a reason for the error? acl.conf.xml - sip_profiles/internal.xml With regards, Sirish From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, 26 January 2013 7:09 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SNOM Phones - BEARERCAPABILITY_NOTAUTH error I confess that I haven't seen this specific error before. Could you get the SIP trace of this call and put it on pastebin as well? That might give you the information you need to crack the case. -MC On Thu, Jan 24, 2013 at 10:41 PM, Sirish Masur Mohan > wrote: Hello All, I think I am missing something obvious here, but I am having trouble trying to make calls between two SNOM phones. I have an open ACL for registration and I keep getting BEARERCAPABILITY_NOTAUTH error when I try to make calls. Would appreciate any inputs on how to debug this! Logs are at : http://pastebin.freeswitch.org/20504 With regards, Sirish _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/1aa0994e/attachment-0001.html From avi at avimarcus.net Wed Jan 30 09:39:26 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 30 Jan 2013 08:39:26 +0200 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: <510870A7.3030607@puzzled.xs4all.nl> <4AA7184D-C10E-4ABC-8336-A9BE53357F6C@url.net.au> Message-ID: Didww has great coverage. I think that's the only positive thing I can say about them. I yanked my numbers from them. -Avi (This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.) On Jan 30, 2013 3:14 AM, "Ashley Breeden" wrote: I find DIDww not too bad for ad-hoc DID's all over the world. Not sure if it meets your exact criteria. Check out www.didww.com Ash. On 30/01/2013, at 12:00 PM, Patrick Lists wrote: > On 01/30/20... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/ddc46abf/attachment.html From garmt.noname at gmail.com Wed Jan 30 11:42:08 2013 From: garmt.noname at gmail.com (grmt) Date: Wed, 30 Jan 2013 09:42:08 +0100 Subject: [Freeswitch-users] voices in the recordings are out of sync In-Reply-To: References: <33095823FD21DF429B481B5163264B799F879B1C8F@VMBX102.ihostexchange.net> Message-ID: <009701cdfec5$b17c53e0$1474fba0$@gmail.com> Did you try other voices? Are you sure you have built the latest version of flite? Download and build flite again. I'm pretty sure this was fixed some time ago. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Monday, January 28, 2013 23:58 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] voices in the recordings are out of sync Ok... You should be testing against MASTER... If you are not testing again Master It is not going to get fixed... All patches go there first... If you can duplicate there, then open a ticket. This is the only way these sort of things get fixed On 1/28/13 4:43 PM, "Yungwei Chen" wrote: Tested against the following version today, and the problem is still there. FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git a424765 2013-01-04 15:45:59Z) From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen Sent: Friday, January 25, 2013 5:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] voices in the recordings are out of sync Hi, I did a quick test against freeswitch-1.2.5.3 on CentOS 5.8, but the problem is still there. In the recording, the caller speaks way faster than he did. Here's the version of freeswitch: FreeSWITCH Version 1.2.5.3+git~20121229T001759Z~e04eab7902 (git e04eab7 2012-12-29 00:17:59Z) Here's the set of steps: 1. In dialplan/default/resolvity.xml, add the following extension: 2. Create scripts/test.js with the following content. This js file will read numbers sequentially starting from 0. session.execute("record_session", "/tmp/test.wav"); for (var i=0; i<100; i++) { session.speak("flite", "kal", i+''); } session.hangup(16); 3. reloadxml 4. dial 0009 from a registered SIP client, and then repeat each number you heard. 5. Listen to the recording, /tmp/test.wav. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, December 18, 2012 8:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] voices in the recordings are out of sync Latest version of FreeSWITCH has some updates that may fix this issue. I would update to 1.2.5.3 ASAP. -MC On Tue, Dec 18, 2012 at 1:56 PM, Yungwei Chen wrote: Hi, I found one issue that voices are always out of sync in the recordings. I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed from yum. I am having trouble installing the latest version from source due to an error: Autoconf version 2.62 or higher is required. It would be nice if someone can reproduce this issue against HEAD. Thanks. Here're the steps to reproduce it. The idea is to call a phone number and then bridge to another phone number while the entire session is being recorded. 1. In dialplan/public.xml, make sure you have a dialplan to handle any 10 digit phone numbers. 2. In dialplan/default/main.xml, make sure you have an extension to handle the call in the default context. 3. In sip_profiles/external/gateways.xml, make sure you have a gateway that allows you to make an outbound call. 4. make a call to one of the allowed 10-digit phone numbers in your environment. 5. Once the call is answered, the caller shall start to count from 1 to 60 with some pause after each number. 6. The callee shall repeat each number he/she heard from the caller. 7. You should be able to hear that 2 voices in the recoridng (/tmp/rec.wav) are out of sync. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/fa79908c/attachment.html From andrew at cassidywebservices.co.uk Wed Jan 30 11:56:43 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 30 Jan 2013 08:56:43 +0000 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: <510870A7.3030607@puzzled.xs4all.nl> <4AA7184D-C10E-4ABC-8336-A9BE53357F6C@url.net.au> Message-ID: for the UK: Magrathea if you want large volumes. Sipgate (Magrathea backed) Andrews & Arnold (AQL Backed) There are many others but I've only really had experience with Magrathea, Sipgate and one other who's network is excellent, just their management has a bit of a reputation. On 30 January 2013 06:39, Avi Marcus wrote: > Didww has great coverage. I think that's the only positive thing I can say > about them. I yanked my numbers from them. > > -Avi > (This message was painstakingly thumbed out on my mobile, so apologies for > brevity and errors.) > > On Jan 30, 2013 3:14 AM, "Ashley Breeden" wrote: > > I find DIDww not too bad for ad-hoc DID's all over the world. Not sure if > it meets your exact criteria. Check out www.didww.com > > Ash. > > > On 30/01/2013, at 12:00 PM, Patrick Lists < > freeswitch-list at puzzled.xs4all.nl> wrote: > > > On 01/30/20... > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/02f5b7d2/attachment-0001.html From andrew at cassidywebservices.co.uk Wed Jan 30 12:26:39 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 30 Jan 2013 09:26:39 +0000 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: <510870A7.3030607@puzzled.xs4all.nl> <4AA7184D-C10E-4ABC-8336-A9BE53357F6C@url.net.au> Message-ID: Oh and SureVoip, I haven't actually used them, mind. They have people on this list. I also have a customer that use VoipFone, seem to be ok but rather limited I've found. On 30 January 2013 08:56, Andrew Cassidy wrote: > for the UK: > > Magrathea if you want large volumes. > Sipgate (Magrathea backed) > Andrews & Arnold (AQL Backed) > > There are many others but I've only really had experience with Magrathea, > Sipgate and one other who's network is excellent, just their management has > a bit of a reputation. > > On 30 January 2013 06:39, Avi Marcus wrote: > >> Didww has great coverage. I think that's the only positive thing I can >> say about them. I yanked my numbers from them. >> >> -Avi >> (This message was painstakingly thumbed out on my mobile, so apologies >> for brevity and errors.) >> >> On Jan 30, 2013 3:14 AM, "Ashley Breeden" wrote: >> >> I find DIDww not too bad for ad-hoc DID's all over the world. Not sure >> if it meets your exact criteria. Check out www.didww.com >> >> Ash. >> >> >> On 30/01/2013, at 12:00 PM, Patrick Lists < >> freeswitch-list at puzzled.xs4all.nl> wrote: >> >> > On 01/30/20... >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/d8627e72/attachment.html From andrew at cassidywebservices.co.uk Wed Jan 30 12:27:32 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 30 Jan 2013 09:27:32 +0000 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: References: Message-ID: I guess my main question is do the other modules accept the new postgres:// syntax or do I still need to use odbc for those? > -- > Andrew Cassidy BSc (Hons) MBCS > http://www.jonisdumb.com/ > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/aab5a7f7/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed Jan 30 15:59:28 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 30 Jan 2013 12:59:28 +0000 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: Wow, thank you everyone for all the feedback so far, this really gives a lot of food for thought. Here's a few more comments; Gamma/GBC - We approached Gamma previously (both GBC and Gamma), but their OUTBOUND prices were extremely high and we never got into discussions about inbound only.. their inter-op testing was also quite lengthy and expensive. Although they are not as expensive as BT, I'd say they fit in the same realm as VM and COLT, high costs and complex inter-op, though I can't comment directly. I'd love to go direct with one of the big players, but sadly the upfront costs and inter-op complexity is too much.. maybe when we grow :) Magrathea - Ah yes.. UK numbers only, their initial prices are a little higher than Simwood (at a minimum charge of ?100 and 50p/number), but hands down beats BT for pricing and simplicity. Their channel concurrency limits are fair, and the on-going prices after the minimum charge are very reasonable. Highly reputable company and not heard a single bad thing about them so far. They might not be as big as BT, but the name has just as reputable. Certainly worth a look for sure. DIDWW - Any else able to offer some reviews on this company - specifically relating to US inbound?? I've heard some bad things about their service quality and support. Avi - could you elaborate more on your experiences with them? Flowroute - Any else able to offer some reviews on this company - specifically relating to US inbound?? VoipFone / SureVoip / Andrews & Arnold / SipGate - both more suited for b2b, and not carrier grade services - plus I haven't heard great things about either.. might be fine if you have one or two non important numbers, but like Andrew said, not for large volume. So far, out of all these providers, the two UK DID providers that stand out the most for me are Magrathea and Simwood. However, I'm still yet to find a US DID provider with a similar reputation.. any additional feedback on US DID providers would be much appreciated. Thanks Cal On Tue, Jan 29, 2013 at 8:13 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello everyone, > > I've been recently reviewing several DID providers, and looking for some > feedback. > > I appreciate this isn't totally on topic for FreeSWITCH, but over the last > few months I've found the community feedback to be invaluable - so I'm > hoping it might open up a good discussion. > > I would be really interested to hear other peoples experiences with their > DID providers, as much detail as possible. > > There are several providers we've looked at, but they are either reselling > others services, or do not deliver the service quality we are looking for. > > The criteria I've been looking for is; > > * Able to provide either US or UK numbers > * Exposes an API that allows control of purchasing and routing > * Guarantees premium call quality, good latency, and inbound CLI. > * Does NOT operate on a market/exchange basis > * Trading as a telecoms provider for more than 5 years > * Has a reputable name within the community or industry > > So far, I have only found the following providers that meet this criteria; > > * Hello Telecom - UK numbers only, ITSPA registered, no charges for > minutes, rental or concurrent channels. No minimum charges. Excellent call > quality/latency, no CLI problems, and almost no downtime in the last 6 > months. However, they do not offer a sane purchasing API which makes > automation difficult, and interop with FS has been somewhat painful. > > * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but > still very fair. Currently at inter-op stage, so can't comment further - > but they seem very good so far. > > * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, > difficult to justify unless you already have existing traffic. > > * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over > 9000 pages of inter-op and around ?10k in set up fees. > > Thanks in advance > > Cal > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/9a19e62a/attachment.html From dominic at timedicer.co.uk Wed Jan 30 15:04:34 2013 From: dominic at timedicer.co.uk (Dominic Raferd) Date: Wed, 30 Jan 2013 12:04:34 +0000 Subject: [Freeswitch-users] FSClient question - messaging Message-ID: <51090C52.8050000@timedicer.co.uk> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/c3fc9639/attachment-0001.html From freeswitch-list at puzzled.xs4all.nl Wed Jan 30 16:32:53 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 30 Jan 2013 14:32:53 +0100 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: <51092105.9050303@puzzled.xs4all.nl> On 01/30/2013 01:59 PM, Cal Leeming [Simplicity Media Ltd] wrote: [snip] > However, I'm still yet to find a US DID provider with a similar > reputation.. any additional feedback on US DID providers would be much > appreciated. Did someone mention voip.ms yet? My experience has been positive, they do US & CAN numbers and have an API: http://voip.ms/rates/xml.php Regards, Patrick From asaad2 at gmail.com Wed Jan 30 16:36:41 2013 From: asaad2 at gmail.com (BookBag) Date: Wed, 30 Jan 2013 08:36:41 -0500 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: I use flowroute. Very good provider On Jan 29, 2013 5:38 PM, "Luke Milbourne" wrote: > Hi Cal, > > Have you investigated AQL? > > I use them currently for my DID, not on the type of scale you are talking > about but I have been impressed with their services so far. > > Luke > > On 29 January 2013 20:55, wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: SPA-2102 Gives 486 instead of Call Waiting (Gilad Abada) >> 2. DID providers - any thoughts? (jan 2013) >> (Cal Leeming [Simplicity Media Ltd]) >> 3. Rejecting undesired codecs in bypass media mode >> (Victor Chukalovskiy) >> 4. Re: DID providers - any thoughts? (jan 2013) (Avi Marcus) >> >> >> ---------- Forwarded message ---------- >> From: Gilad Abada >> To: FreeSWITCH Users Help >> Cc: >> Date: Tue, 29 Jan 2013 14:34:47 -0500 >> Subject: Re: [Freeswitch-users] SPA-2102 Gives 486 instead of Call Waiting >> I think if you turn debug and syslog on level 3 it gives more info >> >> Sent from my mobile device. >> >> On Jan 29, 2013, at 2:28 PM, Avi Marcus wrote: >> >> I did, that's how I got the log in my first email... >> Was I supposed to find more than that? >> -Avi >> >> >> On Tue, Jan 29, 2013 at 9:08 PM, Gilad Abada wrote: >> >>> Avi, >>> >>> You can try to enable syslog trough the GUI and see what the spa 2102 os >>> actually doing. I think it's under the system tab you have syslog and debug >>> server. >>> >>> Sent from my mobile device. >>> >>> On Jan 29, 2013, at 2:05 PM, Avi Marcus wrote: >>> >>> Any suggestions on how to figure this out? I'd really appreciate it! >>> >>> -Avi >>> >>> >>> On Mon, Jan 7, 2013 at 3:12 PM, Avi Marcus wrote: >>> >>>> Here's an intersting one that seems to have no results on google or in >>>> debug info: >>>> >>>> I have an SPA-2102 (firmware 5.2.10) that returns 486/Busy Here when a >>>> call comes in to the active line. >>>> >>>> "call waiting" is turned on, so it should be beeping in instead... >>>> >>>> I got these logs off the SPA but it doesn't seem very helpful: (and >>>> there should be more, I think) >>>> >>>> ==> /var/log/syslog <== >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >>>> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 100 Trying >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 486 Busy Here >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >>>> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >>>> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>> >>>> >>>> I saw one suggestion to do a factory reset that fixed the problem, but >>>> that's a bit troublesome at the moment since the client can't >>>> load provisioning files (dumb network.) >>>> >>>> -Avi Marcus >>>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> ---------- Forwarded message ---------- >> From: "Cal Leeming [Simplicity Media Ltd]" < >> cal.leeming at simplicitymedialtd.co.uk> >> To: FreeSWITCH Users Help >> Cc: >> Date: Tue, 29 Jan 2013 20:13:54 +0000 >> Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) >> Hello everyone, >> >> I've been recently reviewing several DID providers, and looking for some >> feedback. >> >> I appreciate this isn't totally on topic for FreeSWITCH, but over the >> last few months I've found the community feedback to be invaluable - so I'm >> hoping it might open up a good discussion. >> >> I would be really interested to hear other peoples experiences with their >> DID providers, as much detail as possible. >> >> There are several providers we've looked at, but they are either >> reselling others services, or do not deliver the service quality we are >> looking for. >> >> The criteria I've been looking for is; >> >> * Able to provide either US or UK numbers >> * Exposes an API that allows control of purchasing and routing >> * Guarantees premium call quality, good latency, and inbound CLI. >> * Does NOT operate on a market/exchange basis >> * Trading as a telecoms provider for more than 5 years >> * Has a reputable name within the community or industry >> >> So far, I have only found the following providers that meet this criteria; >> >> * Hello Telecom - UK numbers only, ITSPA registered, no charges for >> minutes, rental or concurrent channels. No minimum charges. Excellent call >> quality/latency, no CLI problems, and almost no downtime in the last 6 >> months. However, they do not offer a sane purchasing API which makes >> automation difficult, and interop with FS has been somewhat painful. >> >> * Simwood - UK numbers only, ITSPA registered. Slightly more expensive >> but still very fair. Currently at inter-op stage, so can't comment further >> - but they seem very good so far. >> >> * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, >> difficult to justify unless you already have existing traffic. >> >> * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over >> 9000 pages of inter-op and around ?10k in set up fees. >> >> Thanks in advance >> >> Cal >> >> >> ---------- Forwarded message ---------- >> From: Victor Chukalovskiy >> To: FreeSWITCH Users Help >> Cc: >> Date: Tue, 29 Jan 2013 15:21:44 -0500 >> Subject: [Freeswitch-users] Rejecting undesired codecs in bypass media >> mode >> Hello, >> >> *Call scenario:* call comes in with undesired codec only in SDP. (e.g. >> PCMA) >> >> In my sip profile I have: >> inbound-bypass-media="true" >> inbound-codec-negotiation="greedy" >> inbound-codec-prefs="PCMU,G729" >> >> *What happens:* regardless of the codec in SDP, Freeswitch executes >> dialplan and sends the call to the B-leg. >> >> It also happens regardless of wether I use "late" or "early" negotiation >> >> *What I would like to see:* >> FS rejecting the call right away if SDP does not match >> inbound-codec-prefs. Should not even hit the dial-plan. >> >> Any clue on how to achieve this? >> >> Many thanks, >> -Victor >> >> >> ---------- Forwarded message ---------- >> From: Avi Marcus >> To: FreeSWITCH Users Help >> Cc: >> Date: Tue, 29 Jan 2013 22:55:11 +0200 >> Subject: Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013) >> Shouldn't go without mentioning ukddi.com... >> It's free, since it's subsidized by incoming traffic compensation. He >> claims that's more profitable than termination service is, easily allowing >> him to keep the business running. >> I've not seen the API, I think it has to be specially activated. >> >> He's very responsive to emails. >> >> -Avi >> >> >> On Tue, Jan 29, 2013 at 10:13 PM, Cal Leeming [Simplicity Media Ltd] < >> cal.leeming at simplicitymedialtd.co.uk> wrote: >> >>> Hello everyone, >>> >>> I've been recently reviewing several DID providers, and looking for some >>> feedback. >>> >>> I appreciate this isn't totally on topic for FreeSWITCH, but over the >>> last few months I've found the community feedback to be invaluable - so I'm >>> hoping it might open up a good discussion. >>> >>> I would be really interested to hear other peoples experiences with >>> their DID providers, as much detail as possible. >>> >>> There are several providers we've looked at, but they are either >>> reselling others services, or do not deliver the service quality we are >>> looking for. >>> >>> The criteria I've been looking for is; >>> >>> * Able to provide either US or UK numbers >>> * Exposes an API that allows control of purchasing and routing >>> * Guarantees premium call quality, good latency, and inbound CLI. >>> * Does NOT operate on a market/exchange basis >>> * Trading as a telecoms provider for more than 5 years >>> * Has a reputable name within the community or industry >>> >>> So far, I have only found the following providers that meet this >>> criteria; >>> >>> * Hello Telecom - UK numbers only, ITSPA registered, no charges for >>> minutes, rental or concurrent channels. No minimum charges. Excellent call >>> quality/latency, no CLI problems, and almost no downtime in the last 6 >>> months. However, they do not offer a sane purchasing API which makes >>> automation difficult, and interop with FS has been somewhat painful. >>> >>> * Simwood - UK numbers only, ITSPA registered. Slightly more expensive >>> but still very fair. Currently at inter-op stage, so can't comment further >>> - but they seem very good so far. >>> >>> * Voxbone - Most numbers, ITSPA registered. Extremely high minimum >>> spend, difficult to justify unless you already have existing traffic. >>> >>> * BT - UK numbers only (I think), ITSPA registered (of course lol).. >>> Over 9000 pages of inter-op and around ?10k in set up fees. >>> >>> Thanks in advance >>> >>> Cal >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Luke Milbourne > Email: luke.milbourne at gmail.com > Tel: 07857154817 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/6f72520f/attachment-0001.html From andrew at cassidywebservices.co.uk Wed Jan 30 17:13:24 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 30 Jan 2013 14:13:24 +0000 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: The other thing to remember is many of the UK providers use BT's IPEX platform, and basically bulk-break for you. The one issue I've had with Magrathea is that they don't like it when you make many API calls in quick succession. I once needed to set up 26,000 numbers in bulk and they don't seem to have a better way of doing so but they still complained anyway. On 30 January 2013 13:36, BookBag wrote: > I use flowroute. Very good provider > On Jan 29, 2013 5:38 PM, "Luke Milbourne" > wrote: > >> Hi Cal, >> >> Have you investigated AQL? >> >> I use them currently for my DID, not on the type of scale you are talking >> about but I have been impressed with their services so far. >> >> Luke >> >> On 29 January 2013 20:55, wrote: >> >>> Send FreeSWITCH-users mailing list submissions to >>> freeswitch-users at lists.freeswitch.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> or, via email, send a message with subject or body 'help' to >>> freeswitch-users-request at lists.freeswitch.org >>> >>> You can reach the person managing the list at >>> freeswitch-users-owner at lists.freeswitch.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of FreeSWITCH-users digest..." >>> >>> Today's Topics: >>> >>> 1. Re: SPA-2102 Gives 486 instead of Call Waiting (Gilad Abada) >>> 2. DID providers - any thoughts? (jan 2013) >>> (Cal Leeming [Simplicity Media Ltd]) >>> 3. Rejecting undesired codecs in bypass media mode >>> (Victor Chukalovskiy) >>> 4. Re: DID providers - any thoughts? (jan 2013) (Avi Marcus) >>> >>> >>> ---------- Forwarded message ---------- >>> From: Gilad Abada >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Tue, 29 Jan 2013 14:34:47 -0500 >>> Subject: Re: [Freeswitch-users] SPA-2102 Gives 486 instead of Call >>> Waiting >>> I think if you turn debug and syslog on level 3 it gives more info >>> >>> Sent from my mobile device. >>> >>> On Jan 29, 2013, at 2:28 PM, Avi Marcus wrote: >>> >>> I did, that's how I got the log in my first email... >>> Was I supposed to find more than that? >>> -Avi >>> >>> >>> On Tue, Jan 29, 2013 at 9:08 PM, Gilad Abada wrote: >>> >>>> Avi, >>>> >>>> You can try to enable syslog trough the GUI and see what the spa 2102 >>>> os actually doing. I think it's under the system tab you have syslog and >>>> debug server. >>>> >>>> Sent from my mobile device. >>>> >>>> On Jan 29, 2013, at 2:05 PM, Avi Marcus wrote: >>>> >>>> Any suggestions on how to figure this out? I'd really appreciate it! >>>> >>>> -Avi >>>> >>>> >>>> On Mon, Jan 7, 2013 at 3:12 PM, Avi Marcus wrote: >>>> >>>>> Here's an intersting one that seems to have no results on google or in >>>>> debug info: >>>>> >>>>> I have an SPA-2102 (firmware 5.2.10) that returns 486/Busy Here when a >>>>> call comes in to the active line. >>>>> >>>>> "call waiting" is turned on, so it should be beeping in instead... >>>>> >>>>> I got these logs off the SPA but it doesn't seem very helpful: (and >>>>> there should be more, I think) >>>>> >>>>> ==> /var/log/syslog <== >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >>>>> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>>> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 100 Trying >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 486 Busy Here >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>>> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >>>>> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >>>>> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>>> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>>> >>>>> >>>>> I saw one suggestion to do a factory reset that fixed the problem, but >>>>> that's a bit troublesome at the moment since the client can't >>>>> load provisioning files (dumb network.) >>>>> >>>>> -Avi Marcus >>>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: "Cal Leeming [Simplicity Media Ltd]" < >>> cal.leeming at simplicitymedialtd.co.uk> >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Tue, 29 Jan 2013 20:13:54 +0000 >>> Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) >>> Hello everyone, >>> >>> I've been recently reviewing several DID providers, and looking for some >>> feedback. >>> >>> I appreciate this isn't totally on topic for FreeSWITCH, but over the >>> last few months I've found the community feedback to be invaluable - so I'm >>> hoping it might open up a good discussion. >>> >>> I would be really interested to hear other peoples experiences with >>> their DID providers, as much detail as possible. >>> >>> There are several providers we've looked at, but they are either >>> reselling others services, or do not deliver the service quality we are >>> looking for. >>> >>> The criteria I've been looking for is; >>> >>> * Able to provide either US or UK numbers >>> * Exposes an API that allows control of purchasing and routing >>> * Guarantees premium call quality, good latency, and inbound CLI. >>> * Does NOT operate on a market/exchange basis >>> * Trading as a telecoms provider for more than 5 years >>> * Has a reputable name within the community or industry >>> >>> So far, I have only found the following providers that meet this >>> criteria; >>> >>> * Hello Telecom - UK numbers only, ITSPA registered, no charges for >>> minutes, rental or concurrent channels. No minimum charges. Excellent call >>> quality/latency, no CLI problems, and almost no downtime in the last 6 >>> months. However, they do not offer a sane purchasing API which makes >>> automation difficult, and interop with FS has been somewhat painful. >>> >>> * Simwood - UK numbers only, ITSPA registered. Slightly more expensive >>> but still very fair. Currently at inter-op stage, so can't comment further >>> - but they seem very good so far. >>> >>> * Voxbone - Most numbers, ITSPA registered. Extremely high minimum >>> spend, difficult to justify unless you already have existing traffic. >>> >>> * BT - UK numbers only (I think), ITSPA registered (of course lol).. >>> Over 9000 pages of inter-op and around ?10k in set up fees. >>> >>> Thanks in advance >>> >>> Cal >>> >>> >>> ---------- Forwarded message ---------- >>> From: Victor Chukalovskiy >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Tue, 29 Jan 2013 15:21:44 -0500 >>> Subject: [Freeswitch-users] Rejecting undesired codecs in bypass media >>> mode >>> Hello, >>> >>> *Call scenario:* call comes in with undesired codec only in SDP. (e.g. >>> PCMA) >>> >>> In my sip profile I have: >>> inbound-bypass-media="true" >>> inbound-codec-negotiation="greedy" >>> inbound-codec-prefs="PCMU,G729" >>> >>> *What happens:* regardless of the codec in SDP, Freeswitch executes >>> dialplan and sends the call to the B-leg. >>> >>> It also happens regardless of wether I use "late" or "early" negotiation >>> >>> *What I would like to see:* >>> FS rejecting the call right away if SDP does not match >>> inbound-codec-prefs. Should not even hit the dial-plan. >>> >>> Any clue on how to achieve this? >>> >>> Many thanks, >>> -Victor >>> >>> >>> ---------- Forwarded message ---------- >>> From: Avi Marcus >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Tue, 29 Jan 2013 22:55:11 +0200 >>> Subject: Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013) >>> Shouldn't go without mentioning ukddi.com... >>> It's free, since it's subsidized by incoming traffic compensation. He >>> claims that's more profitable than termination service is, easily allowing >>> him to keep the business running. >>> I've not seen the API, I think it has to be specially activated. >>> >>> He's very responsive to emails. >>> >>> -Avi >>> >>> >>> On Tue, Jan 29, 2013 at 10:13 PM, Cal Leeming [Simplicity Media Ltd] < >>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>> >>>> Hello everyone, >>>> >>>> I've been recently reviewing several DID providers, and looking for >>>> some feedback. >>>> >>>> I appreciate this isn't totally on topic for FreeSWITCH, but over the >>>> last few months I've found the community feedback to be invaluable - so I'm >>>> hoping it might open up a good discussion. >>>> >>>> I would be really interested to hear other peoples experiences with >>>> their DID providers, as much detail as possible. >>>> >>>> There are several providers we've looked at, but they are either >>>> reselling others services, or do not deliver the service quality we are >>>> looking for. >>>> >>>> The criteria I've been looking for is; >>>> >>>> * Able to provide either US or UK numbers >>>> * Exposes an API that allows control of purchasing and routing >>>> * Guarantees premium call quality, good latency, and inbound CLI. >>>> * Does NOT operate on a market/exchange basis >>>> * Trading as a telecoms provider for more than 5 years >>>> * Has a reputable name within the community or industry >>>> >>>> So far, I have only found the following providers that meet this >>>> criteria; >>>> >>>> * Hello Telecom - UK numbers only, ITSPA registered, no charges for >>>> minutes, rental or concurrent channels. No minimum charges. Excellent call >>>> quality/latency, no CLI problems, and almost no downtime in the last 6 >>>> months. However, they do not offer a sane purchasing API which makes >>>> automation difficult, and interop with FS has been somewhat painful. >>>> >>>> * Simwood - UK numbers only, ITSPA registered. Slightly more expensive >>>> but still very fair. Currently at inter-op stage, so can't comment further >>>> - but they seem very good so far. >>>> >>>> * Voxbone - Most numbers, ITSPA registered. Extremely high minimum >>>> spend, difficult to justify unless you already have existing traffic. >>>> >>>> * BT - UK numbers only (I think), ITSPA registered (of course lol).. >>>> Over 9000 pages of inter-op and around ?10k in set up fees. >>>> >>>> Thanks in advance >>>> >>>> Cal >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Luke Milbourne >> Email: luke.milbourne at gmail.com >> Tel: 07857154817 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/835db73a/attachment-0001.html From frederick at targointernet.com Wed Jan 30 17:18:34 2013 From: frederick at targointernet.com (Frederick Pruneau) Date: Wed, 30 Jan 2013 09:18:34 -0500 Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls Message-ID: <51092BBA.8030001@targointernet.com> Hi guys! I have a problem with my freeswitch server and I need some help. All incoming calls are not working since yesterday. All I did is to change the IP address. The only error message I have is this one: [WARNING] switch_core_state_machine.c:514 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 Abandoned [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] What does Wrong call state mean and what can I do to fix this issue? Everything else is working. I can make outgoing calls and call other extensions. Thanks in advance. Fred From mitch.capper at gmail.com Wed Jan 30 18:35:44 2013 From: mitch.capper at gmail.com (Mitch Capper) Date: Wed, 30 Jan 2013 07:35:44 -0800 Subject: [Freeswitch-users] FSClient question - messaging In-Reply-To: <51090C52.8050000@timedicer.co.uk> References: <51090C52.8050000@timedicer.co.uk> Message-ID: Hi Dominic, I am happy you like FSClient. It is built ontop of freeswitch and freeswitch does support text messaging so its mainly just the interface that is needed. I have not heard a lot of demand for messaging support and its not something I personally have great need for. Given the fact the core supports it however it may not be a large project to add it on. If you have some programming experience you may try yourself. Otherwise I do not have plans to add it in the near term future but will add it to the todo list. ~Mitch On Wed, Jan 30, 2013 at 4:04 AM, Dominic Raferd wrote: > Hi, I'm impressed by FSClient's ability to work with the controls on the > Jabra and Plantronics headsets, and would like to use it. But it doesn't > seem to have any built-in text messaging capability unlike most other SIP > clients e.g. MicroSIP. > > Is there a way to get text messaging working with FSClient, or is it a > capability that anyone has thought of adding? > > Thanks and regards > > Dominic > -- > *TimeDicer* : Free File Recovery from Whenever > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/e3a12703/attachment.html From alessio at asistar.it Wed Jan 30 20:06:39 2013 From: alessio at asistar.it (Alessio) Date: Wed, 30 Jan 2013 18:06:39 +0100 Subject: [Freeswitch-users] Patton FXO configuration on Freeswitch - more details In-Reply-To: <510790C9.204@asistar.it> References: <5102AAF6.2070201@asistar.it> <51062C9C.8020701@asistar.it> <510790C9.204@asistar.it> Message-ID: <5109531F.1080807@asistar.it> Nella citazione in data marted? 29 gennaio 2013 10:05:13, Alessio ha scritto: > Nella citazione in data luned? 28 gennaio 2013 19:43:12, Michael > Collins ha scritto: >> >> Hi Michael, >> thanks for the reply. >> >> So if I register in directory as a phone how do I get out of the >> gateway, for example through a bridge? >> >> Alessio >> >> >> User directory is for inbound registrations and SIP auth challenges >> for devices connecting to the CudaTel. Gateways are the opposite - >> they are for auth challenges and registrations going from the CudaTel >> to other devices. To dial out a gateway you just use the syntax: >> >> > data="sofia/gateway/gateway_name/${phone_number_to_send_to_gateway}"/> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > So for incoming calls and recording to the PBX should I configure FXO > in conf/directory. > To make outgoing calls through the FXO I configure a gateway in > conf/sip_profile. > > Is that correct? > > If so, how do I configure the gateway? > > Thank you, > Alessio Hi all! I've solved! For the gateway registration you need to configure a user: And to make outbound call you need to configure an internal gateway: The FXO works fine. Thanks! From msc at freeswitch.org Wed Jan 30 20:09:52 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 Jan 2013 09:09:52 -0800 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Today Message-ID: Hello folks, Our conference call starts in less than an hour. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_30 Today we are going to do a refresher on event socket and ESL, specifically on the commands you can do when controlling a phone call. We are finding more and more people are using event socket and ESL and some of the same questions are coming up so we thought it would be a good idea to review a few things when it comes to using the event socket. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/cac6e29a/attachment.html From anthony.minessale at gmail.com Thu Jan 31 04:07:39 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 Jan 2013 19:07:39 -0600 Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls In-Reply-To: <51092BBA.8030001@targointernet.com> References: <51092BBA.8030001@targointernet.com> Message-ID: That means when the call was challenged for auth, it never replied with the new INVITE with www-auth header. You may have some network settings wrong making the followup message go to the wrong host. On Wed, Jan 30, 2013 at 8:18 AM, Frederick Pruneau < frederick at targointernet.com> wrote: > Hi guys! > > I have a problem with my freeswitch server and I need some help. All > incoming calls are not working since yesterday. All I did is to change > the IP address. The only error message I have is this one: > > [WARNING] switch_core_state_machine.c:514 > 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb > sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 Abandoned > [NOTICE] switch_core_state_machine.c:517 Hangup > sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] > > What does Wrong call state mean and what can I do to fix this issue? > > Everything else is working. I can make outgoing calls and call other > extensions. > > Thanks in advance. > > Fred > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130130/388c1835/attachment.html From cal.leeming at simplicitymedialtd.co.uk Thu Jan 31 04:12:43 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 31 Jan 2013 01:12:43 +0000 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: I think ideally, these are the kind of providers we want to stick with - otherwise you end up going through a reseller of a reseller of a reseller of a reseller.. not nice. It's a shame their API doesn't support bulk operations, but as long as they don't have rate limit police like Twitter, then I can live with this :) Cal On Wed, Jan 30, 2013 at 2:13 PM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > The other thing to remember is many of the UK providers use BT's IPEX > platform, and basically bulk-break for you. > > The one issue I've had with Magrathea is that they don't like it when you > make many API calls in quick succession. I once needed to set up 26,000 > numbers in bulk and they don't seem to have a better way of doing so but > they still complained anyway. > > > On 30 January 2013 13:36, BookBag wrote: > >> I use flowroute. Very good provider >> On Jan 29, 2013 5:38 PM, "Luke Milbourne" >> wrote: >> >>> Hi Cal, >>> >>> Have you investigated AQL? >>> >>> I use them currently for my DID, not on the type of scale you are >>> talking about but I have been impressed with their services so far. >>> >>> Luke >>> >>> On 29 January 2013 20:55, >> > wrote: >>> >>>> Send FreeSWITCH-users mailing list submissions to >>>> freeswitch-users at lists.freeswitch.org >>>> >>>> To subscribe or unsubscribe via the World Wide Web, visit >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> or, via email, send a message with subject or body 'help' to >>>> freeswitch-users-request at lists.freeswitch.org >>>> >>>> You can reach the person managing the list at >>>> freeswitch-users-owner at lists.freeswitch.org >>>> >>>> When replying, please edit your Subject line so it is more specific >>>> than "Re: Contents of FreeSWITCH-users digest..." >>>> >>>> Today's Topics: >>>> >>>> 1. Re: SPA-2102 Gives 486 instead of Call Waiting (Gilad Abada) >>>> 2. DID providers - any thoughts? (jan 2013) >>>> (Cal Leeming [Simplicity Media Ltd]) >>>> 3. Rejecting undesired codecs in bypass media mode >>>> (Victor Chukalovskiy) >>>> 4. Re: DID providers - any thoughts? (jan 2013) (Avi Marcus) >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Gilad Abada >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Tue, 29 Jan 2013 14:34:47 -0500 >>>> Subject: Re: [Freeswitch-users] SPA-2102 Gives 486 instead of Call >>>> Waiting >>>> I think if you turn debug and syslog on level 3 it gives more info >>>> >>>> Sent from my mobile device. >>>> >>>> On Jan 29, 2013, at 2:28 PM, Avi Marcus wrote: >>>> >>>> I did, that's how I got the log in my first email... >>>> Was I supposed to find more than that? >>>> -Avi >>>> >>>> >>>> On Tue, Jan 29, 2013 at 9:08 PM, Gilad Abada wrote: >>>> >>>>> Avi, >>>>> >>>>> You can try to enable syslog trough the GUI and see what the spa 2102 >>>>> os actually doing. I think it's under the system tab you have syslog and >>>>> debug server. >>>>> >>>>> Sent from my mobile device. >>>>> >>>>> On Jan 29, 2013, at 2:05 PM, Avi Marcus wrote: >>>>> >>>>> Any suggestions on how to figure this out? I'd really appreciate it! >>>>> >>>>> -Avi >>>>> >>>>> >>>>> On Mon, Jan 7, 2013 at 3:12 PM, Avi Marcus wrote: >>>>> >>>>>> Here's an intersting one that seems to have no results on google or >>>>>> in debug info: >>>>>> >>>>>> I have an SPA-2102 (firmware 5.2.10) that returns 486/Busy Here when >>>>>> a call comes in to the active line. >>>>>> >>>>>> "call waiting" is turned on, so it should be beeping in instead... >>>>>> >>>>>> I got these logs off the SPA but it doesn't seem very helpful: (and >>>>>> there should be more, I think) >>>>>> >>>>>> ==> /var/log/syslog <== >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(1148) >>>>>> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>>>> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(312) >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 100 Trying >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il[0]->$FreeSWITCH: 5060(338) >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.ilSIP/2.0 486 Busy Here >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>>>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il#012 >>>>>> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >>>>>> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il[0]<<$FreeSWITCH: 5060(334) >>>>>> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>>>> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>>>>> >>>>>> >>>>>> I saw one suggestion to do a factory reset that fixed the problem, >>>>>> but that's a bit troublesome at the moment since the client can't >>>>>> load provisioning files (dumb network.) >>>>>> >>>>>> -Avi Marcus >>>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: "Cal Leeming [Simplicity Media Ltd]" < >>>> cal.leeming at simplicitymedialtd.co.uk> >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Tue, 29 Jan 2013 20:13:54 +0000 >>>> Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) >>>> Hello everyone, >>>> >>>> I've been recently reviewing several DID providers, and looking for >>>> some feedback. >>>> >>>> I appreciate this isn't totally on topic for FreeSWITCH, but over the >>>> last few months I've found the community feedback to be invaluable - so I'm >>>> hoping it might open up a good discussion. >>>> >>>> I would be really interested to hear other peoples experiences with >>>> their DID providers, as much detail as possible. >>>> >>>> There are several providers we've looked at, but they are either >>>> reselling others services, or do not deliver the service quality we are >>>> looking for. >>>> >>>> The criteria I've been looking for is; >>>> >>>> * Able to provide either US or UK numbers >>>> * Exposes an API that allows control of purchasing and routing >>>> * Guarantees premium call quality, good latency, and inbound CLI. >>>> * Does NOT operate on a market/exchange basis >>>> * Trading as a telecoms provider for more than 5 years >>>> * Has a reputable name within the community or industry >>>> >>>> So far, I have only found the following providers that meet this >>>> criteria; >>>> >>>> * Hello Telecom - UK numbers only, ITSPA registered, no charges for >>>> minutes, rental or concurrent channels. No minimum charges. Excellent call >>>> quality/latency, no CLI problems, and almost no downtime in the last 6 >>>> months. However, they do not offer a sane purchasing API which makes >>>> automation difficult, and interop with FS has been somewhat painful. >>>> >>>> * Simwood - UK numbers only, ITSPA registered. Slightly more expensive >>>> but still very fair. Currently at inter-op stage, so can't comment further >>>> - but they seem very good so far. >>>> >>>> * Voxbone - Most numbers, ITSPA registered. Extremely high minimum >>>> spend, difficult to justify unless you already have existing traffic. >>>> >>>> * BT - UK numbers only (I think), ITSPA registered (of course lol).. >>>> Over 9000 pages of inter-op and around ?10k in set up fees. >>>> >>>> Thanks in advance >>>> >>>> Cal >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Victor Chukalovskiy >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Tue, 29 Jan 2013 15:21:44 -0500 >>>> Subject: [Freeswitch-users] Rejecting undesired codecs in bypass media >>>> mode >>>> Hello, >>>> >>>> *Call scenario:* call comes in with undesired codec only in SDP. (e.g. >>>> PCMA) >>>> >>>> In my sip profile I have: >>>> inbound-bypass-media="true" >>>> inbound-codec-negotiation="greedy" >>>> inbound-codec-prefs="PCMU,G729" >>>> >>>> *What happens:* regardless of the codec in SDP, Freeswitch executes >>>> dialplan and sends the call to the B-leg. >>>> >>>> It also happens regardless of wether I use "late" or "early" negotiation >>>> >>>> *What I would like to see:* >>>> FS rejecting the call right away if SDP does not match >>>> inbound-codec-prefs. Should not even hit the dial-plan. >>>> >>>> Any clue on how to achieve this? >>>> >>>> Many thanks, >>>> -Victor >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Avi Marcus >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Tue, 29 Jan 2013 22:55:11 +0200 >>>> Subject: Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013) >>>> Shouldn't go without mentioning ukddi.com... >>>> It's free, since it's subsidized by incoming traffic compensation. He >>>> claims that's more profitable than termination service is, easily allowing >>>> him to keep the business running. >>>> I've not seen the API, I think it has to be specially activated. >>>> >>>> He's very responsive to emails. >>>> >>>> -Avi >>>> >>>> >>>> On Tue, Jan 29, 2013 at 10:13 PM, Cal Leeming [Simplicity Media Ltd] < >>>> cal.leeming at simplicitymedialtd.co.uk> wrote: >>>> >>>>> Hello everyone, >>>>> >>>>> I've been recently reviewing several DID providers, and looking for >>>>> some feedback. >>>>> >>>>> I appreciate this isn't totally on topic for FreeSWITCH, but over the >>>>> last few months I've found the community feedback to be invaluable - so I'm >>>>> hoping it might open up a good discussion. >>>>> >>>>> I would be really interested to hear other peoples experiences with >>>>> their DID providers, as much detail as possible. >>>>> >>>>> There are several providers we've looked at, but they are either >>>>> reselling others services, or do not deliver the service quality we are >>>>> looking for. >>>>> >>>>> The criteria I've been looking for is; >>>>> >>>>> * Able to provide either US or UK numbers >>>>> * Exposes an API that allows control of purchasing and routing >>>>> * Guarantees premium call quality, good latency, and inbound CLI. >>>>> * Does NOT operate on a market/exchange basis >>>>> * Trading as a telecoms provider for more than 5 years >>>>> * Has a reputable name within the community or industry >>>>> >>>>> So far, I have only found the following providers that meet this >>>>> criteria; >>>>> >>>>> * Hello Telecom - UK numbers only, ITSPA registered, no charges for >>>>> minutes, rental or concurrent channels. No minimum charges. Excellent call >>>>> quality/latency, no CLI problems, and almost no downtime in the last 6 >>>>> months. However, they do not offer a sane purchasing API which makes >>>>> automation difficult, and interop with FS has been somewhat painful. >>>>> >>>>> * Simwood - UK numbers only, ITSPA registered. Slightly more expensive >>>>> but still very fair. Currently at inter-op stage, so can't comment further >>>>> - but they seem very good so far. >>>>> >>>>> * Voxbone - Most numbers, ITSPA registered. Extremely high minimum >>>>> spend, difficult to justify unless you already have existing traffic. >>>>> >>>>> * BT - UK numbers only (I think), ITSPA registered (of course lol).. >>>>> Over 9000 pages of inter-op and around ?10k in set up fees. >>>>> >>>>> Thanks in advance >>>>> >>>>> Cal >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Luke Milbourne >>> Email: luke.milbourne at gmail.com >>> Tel: 07857154817 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/1625e75f/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Thu Jan 31 04:39:17 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 31 Jan 2013 01:39:17 +0000 Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) In-Reply-To: References: Message-ID: Just been looking over a few other providers.. VoxBeam look really good, but sadly they do not offer an automation API yet - which is a real kicker for us :( Conexiant also seem good, but they charge per minute and don't list their concurrent channel limits.. is this a common thing among US DID providers?? DIDnumbers seems to have a great range of numbers, good prices, but doesn't seem to have any notable reputation. Out of all of them, Conexiant appear to be the most reputable for US DIDs.. any thoughts??? Cal On Tue, Jan 29, 2013 at 8:13 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello everyone, > > I've been recently reviewing several DID providers, and looking for some > feedback. > > I appreciate this isn't totally on topic for FreeSWITCH, but over the last > few months I've found the community feedback to be invaluable - so I'm > hoping it might open up a good discussion. > > I would be really interested to hear other peoples experiences with their > DID providers, as much detail as possible. > > There are several providers we've looked at, but they are either reselling > others services, or do not deliver the service quality we are looking for. > > The criteria I've been looking for is; > > * Able to provide either US or UK numbers > * Exposes an API that allows control of purchasing and routing > * Guarantees premium call quality, good latency, and inbound CLI. > * Does NOT operate on a market/exchange basis > * Trading as a telecoms provider for more than 5 years > * Has a reputable name within the community or industry > > So far, I have only found the following providers that meet this criteria; > > * Hello Telecom - UK numbers only, ITSPA registered, no charges for > minutes, rental or concurrent channels. No minimum charges. Excellent call > quality/latency, no CLI problems, and almost no downtime in the last 6 > months. However, they do not offer a sane purchasing API which makes > automation difficult, and interop with FS has been somewhat painful. > > * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but > still very fair. Currently at inter-op stage, so can't comment further - > but they seem very good so far. > > * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, > difficult to justify unless you already have existing traffic. > > * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over > 9000 pages of inter-op and around ?10k in set up fees. > > Thanks in advance > > Cal > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/3e59b563/attachment.html From gabe at gundy.org Thu Jan 31 05:30:24 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 30 Jan 2013 19:30:24 -0700 Subject: [Freeswitch-users] Two Freeswitches - Invalid Gateway In-Reply-To: <5101CA55.9040006@xmission.com> References: <5101CA55.9040006@xmission.com> Message-ID: On Thu, Jan 24, 2013 at 4:57 PM, Pete Ashdown wrote: > The result is that I get an "Invalid Gateway" when trying to dial > extensions on either box, like it is trying to go through the SIP > external profile. Any thoughts as to what I'm doing wrong, or what is > missing from the above configuration? Can you share the snippet of the dialplan that shows the dialstring your using? Also, you might tell us a bit about your sofia profiles. Best, Gabe --happy user of xmisson's mirrors ;) From toddb at toddbailey.net Thu Jan 31 08:49:56 2013 From: toddb at toddbailey.net (Todd Bailey) Date: Wed, 30 Jan 2013 21:49:56 -0800 Subject: [Freeswitch-users] Low Volume levels on ip phones In-Reply-To: <1341853685.4860.3.camel@mythtv> References: <1341685164.3454.51.camel@mythtv> <1341748446.53554.YahooMailNeo@web39304.mail.mud.yahoo.com> <1341812640.3767.6.camel@mythtv> <1341823757.69136.YahooMailNeo@web39303.mail.mud.yahoo.com> <1341853685.4860.3.camel@mythtv> Message-ID: <1359611396.16353.95.camel@mythtv.toddbailey.net> Hello All, Is there an easy way to boost volume levels going to a ip phone? I have the volume set to max and the volume levels is still very quiet? I already adjusted the ata a spa3102 to boost the pstn to spa gain to max which helped a little but not by much The primary phones are polycom 501's thanks From cmason at frontiernetworks.ca Thu Jan 31 12:40:29 2013 From: cmason at frontiernetworks.ca (Colin Mason) Date: Thu, 31 Jan 2013 04:40:29 -0500 Subject: [Freeswitch-users] Limit calls per user and transferring issue Message-ID: <0D1C698866F66045A6201FD0F59CAC900145DC8D32@EX.frontier.local> Hello, I am running 1.3.13b git 8859eb0 2012-12-30 19:08:55Z I am trying to use limit to keep the maximum calls (in+out) per registration at one. I am using 2 Linksys SPA303 phones both registered as extension 1000. I am using BLF and it works. I can transfer a call to the other phone by pressing hold on phone 1 and pressing the blinking BLF light on phone 2. I am using limit_ignore_transfer=true. If I transfer an inbound call to the other phone, the limit for that user persists and everything works. If I transfer an outbound call over to the other phone, the limit decrements to 0 which is unintended in my setup. I was wondering if there was a way to make transferring outbound calls to another phone have the same limit behavior as transferring inbound calls. I read that this can be done using loopbacks but I am unsure how to approach using a loopback to perform this function. Any help is much appreciated. Colin From vipkilla at gmail.com Thu Jan 31 16:36:13 2013 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 31 Jan 2013 08:36:13 -0500 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: References: Message-ID: It depends if the modules have been updated to use the syntax. If you look in the GIT log you can see where it's been applied. AFAIK it's been applied to all modules EXCEPT mod_nibblebill. I wrote a patch for mod_nibblebill about 3 months ago to use the pgsql syntax but it has been ignored, the patch is on jira. On Wed, Jan 30, 2013 at 4:27 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > I guess my main question is do the other modules accept the new > postgres:// syntax or do I still need to use odbc for those? > >> -- >> Andrew Cassidy BSc (Hons) MBCS >> http://www.jonisdumb.com/ >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/f6a0a96d/attachment.html From steveayre at gmail.com Thu Jan 31 17:29:38 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 31 Jan 2013 14:29:38 +0000 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: References: Message-ID: They should, it's implemented within the core API that the modules use so doesn't require every module to reimplement the postgresql support themselves. Rather the modules should no longer do the verification of the DSN themselves and should be connecting using the new core API function - as long as they've been updated for that then they should work and any future changes to the DSN formats shouldn't require any changes in the modules. Any not updated yet should be - if you find any such while testing raise a Jira on them (http://jira.freeswitch.org). -Steve On 30 January 2013 09:27, Andrew Cassidy wrote: > I guess my main question is do the other modules accept the new > postgres:// syntax or do I still need to use odbc for those? > >> -- >> Andrew Cassidy BSc (Hons) MBCS >> http://www.jonisdumb.com/ >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/f13003e6/attachment-0001.html From vipkilla at gmail.com Thu Jan 31 17:55:19 2013 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 31 Jan 2013 09:55:19 -0500 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: References: Message-ID: > Any not updated yet should be - if you find any such while testing raise a > Jira on them (http://jira.freeswitch.org). > ^^^ mod_nibblebill needs to be updated... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/3b16a823/attachment.html From frederick at targointernet.com Thu Jan 31 17:57:32 2013 From: frederick at targointernet.com (Frederick Pruneau) Date: Thu, 31 Jan 2013 09:57:32 -0500 Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls In-Reply-To: References: <51092BBA.8030001@targointernet.com> Message-ID: <510A865C.505@targointernet.com> Hi Anthony, First of all, thanks for your reply! I appreciate your help! I checked conf files that could have changed since the new ip address has been set. I tried to rollback as far as I can but did not find the option that makes incoming calls not working. I checked also my network settings. Everything seems to be good. I'm probably missing something in freeswitch config files. Is there a file (or files) that I need to modify and put my new IP? Thanks again. Fred Le 2013-01-30 20:07, Anthony Minessale a ?crit : > That means when the call was challenged for auth, it never replied > with the new INVITE with www-auth header. > You may have some network settings wrong making the followup message > go to the wrong host. > > > > On Wed, Jan 30, 2013 at 8:18 AM, Frederick Pruneau > > wrote: > > Hi guys! > > I have a problem with my freeswitch server and I need some help. All > incoming calls are not working since yesterday. All I did is to change > the IP address. The only error message I have is this one: > > [WARNING] switch_core_state_machine.c:514 > 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb > sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 Abandoned > [NOTICE] switch_core_state_machine.c:517 Hangup > sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] > > What does Wrong call state mean and what can I do to fix this issue? > > Everything else is working. I can make outgoing calls and call other > extensions. > > Thanks in advance. > > Fred > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/7cb28668/attachment.html From andrew at cassidywebservices.co.uk Thu Jan 31 18:26:55 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 31 Jan 2013 15:26:55 +0000 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: References: Message-ID: Thanks, I'll give it a go later! On 31 January 2013 14:55, Vik Killa wrote: > > Any not updated yet should be - if you find any such while testing raise a >> Jira on them (http://jira.freeswitch.org). >> > > ^^^ mod_nibblebill needs to be updated... > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/1d15a533/attachment.html From anthony.minessale at gmail.com Thu Jan 31 19:15:21 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Jan 2013 10:15:21 -0600 Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls In-Reply-To: <510A865C.505@targointernet.com> References: <51092BBA.8030001@targointernet.com> <510A865C.505@targointernet.com> Message-ID: Its hard to tell without seeing a trace. You should turn up the console logs and enabled debugging. sofia global siptrace on console loglevel debug On Thu, Jan 31, 2013 at 8:57 AM, Frederick Pruneau < frederick at targointernet.com> wrote: > Hi Anthony, > > First of all, thanks for your reply! I appreciate your help! > > I checked conf files that could have changed since the new ip address has > been set. I tried to rollback as far as I can but did not find the option > that makes incoming calls not working. I checked also my network settings. > Everything seems to be good. I'm probably missing something in freeswitch > config files. > > Is there a file (or files) that I need to modify and put my new IP? > > Thanks again. > > Fred > > Le 2013-01-30 20:07, Anthony Minessale a ?crit : > > That means when the call was challenged for auth, it never replied with > the new INVITE with www-auth header. > You may have some network settings wrong making the followup message go to > the wrong host. > > > > On Wed, Jan 30, 2013 at 8:18 AM, Frederick Pruneau < > frederick at targointernet.com> wrote: > >> Hi guys! >> >> I have a problem with my freeswitch server and I need some help. All >> incoming calls are not working since yesterday. All I did is to change >> the IP address. The only error message I have is this one: >> >> [WARNING] switch_core_state_machine.c:514 >> 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb >> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 Abandoned >> [NOTICE] switch_core_state_machine.c:517 Hangup >> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] >> >> What does Wrong call state mean and what can I do to fix this issue? >> >> Everything else is working. I can make outgoing calls and call other >> extensions. >> >> Thanks in advance. >> >> Fred >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/2bee9c5f/attachment-0001.html From anthony.minessale at gmail.com Thu Jan 31 19:23:28 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Jan 2013 10:23:28 -0600 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: References: Message-ID: Ignored is not really a very positive sentiment.... We have a lot to deal with. So if there is a patch of this nature that is not committed yet, its just a matter of finding time to go over them. Please do not generate negative connotations. Its in tree now and I would appreciate some understanding in the future. On Thu, Jan 31, 2013 at 9:26 AM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > Thanks, I'll give it a go later! > > On 31 January 2013 14:55, Vik Killa wrote: > >> >> Any not updated yet should be - if you find any such while testing raise >>> a Jira on them (http://jira.freeswitch.org). >>> >> >> ^^^ mod_nibblebill needs to be updated... >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/4c651a0a/attachment.html From vipkilla at gmail.com Thu Jan 31 19:35:30 2013 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 31 Jan 2013 11:35:30 -0500 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: References: Message-ID: I apologize for my negative tone. It's just a bit frustrating having patches sit there for months. I understand you are busy and I don't expect you to go over the code right away. It would be nice if there was someone with more time and commit powers that could review some patches. On Thu, Jan 31, 2013 at 11:23 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Ignored is not really a very positive sentiment.... > > We have a lot to deal with. So if there is a patch of this nature that is > not committed yet, its just a matter of finding time to go over them. > Please do not generate negative connotations. > > Its in tree now and I would appreciate some understanding in the future. > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/1fb3d62e/attachment.html From 8f27e956 at gmail.com Thu Jan 31 19:40:59 2013 From: 8f27e956 at gmail.com (S. Scott) Date: Thu, 31 Jan 2013 11:40:59 -0500 Subject: [Freeswitch-users] IAX Message-ID: <-1407881447721643425@unknownmsgid> I've done the google searches and it's not clear what the current status of IAX facility is available in fs. We have a project needing IAX trunk with registration support. With thanks, ????? iThing: Big thumbs & little keys. Please excuse typo, spelling and grammar errors ? Thought of the Day ? "With all this manure, there must be a pony in here somewhere.? From steveayre at gmail.com Thu Jan 31 19:46:57 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 31 Jan 2013 16:46:57 +0000 Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls In-Reply-To: <510A865C.505@targointernet.com> References: <51092BBA.8030001@targointernet.com> <510A865C.505@targointernet.com> Message-ID: <06E0038D-469F-47F1-8449-C6B590AAECA0@gmail.com> Check the sip-ip rtp-ip settings on your Sofia profiles. They may be using a variable such as local_ip4, if so check that too. Did you restart FS or at least the relevant profiles after changing the IP? Note that if you're not setting it explicitly it may be autodetecting your IP, in which case that might not happen correctly if your server now has multiple IPs. Or using the old one if FS wasn't restarted. Also remember Sofia cannot listen on 0.0.0.0 If your FS is behind NAT then that also may imply a few other settings need checking too (ext-sip-ip etc). Steve On 31 Jan 2013, at 14:57, Frederick Pruneau wrote: > Hi Anthony, > > First of all, thanks for your reply! I appreciate your help! > > I checked conf files that could have changed since the new ip address has been set. I tried to rollback as far as I can but did not find the option that makes incoming calls not working. I checked also my network settings. Everything seems to be good. I'm probably missing something in freeswitch config files. > > Is there a file (or files) that I need to modify and put my new IP? > > Thanks again. > > Fred > > Le 2013-01-30 20:07, Anthony Minessale a ?crit : >> That means when the call was challenged for auth, it never replied with the new INVITE with www-auth header. >> You may have some network settings wrong making the followup message go to the wrong host. >> >> >> >> On Wed, Jan 30, 2013 at 8:18 AM, Frederick Pruneau wrote: >>> Hi guys! >>> >>> I have a problem with my freeswitch server and I need some help. All >>> incoming calls are not working since yesterday. All I did is to change >>> the IP address. The only error message I have is this one: >>> >>> [WARNING] switch_core_state_machine.c:514 >>> 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb >>> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 Abandoned >>> [NOTICE] switch_core_state_machine.c:517 Hangup >>> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] >>> >>> What does Wrong call state mean and what can I do to fix this issue? >>> >>> Everything else is working. I can make outgoing calls and call other >>> extensions. >>> >>> Thanks in advance. >>> >>> Fred >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/5d876b9c/attachment-0001.html From anthony.minessale at gmail.com Thu Jan 31 19:48:15 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Jan 2013 10:48:15 -0600 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: References: Message-ID: Try to shift your mentality so you try to get patches to make it into releases more importantly than into HEAD. If the patch still compiles then its not a big deal if its not in yet as long as it cleanly applies and makes it in before a tag. We are separating the release structure to build multiple channels. This is not easy. If we don't do it people will get mad that we have no stable channel. If we do it, people get mad we are too slow to add their patches. Unfortunately someone will always be mad. Even George Lucas gets yelled at for not making Star Wars right. Sadly I must make decisions that are not 100% favorable but your patch is a welcomed addition since we clearly missed it when we changed all the other modules. Going forward think of it terms of seasons. We will have release season where we slow down and make sure the code is stable and we will have development season when we change everything. 1.5 is coming out soon as the HEAD for 1.4. Then we can look at some more patches that have been lingering. On Thu, Jan 31, 2013 at 10:35 AM, Vik Killa wrote: > I apologize for my negative tone. It's just a bit frustrating having > patches sit there for months. I understand you are busy and I don't expect > you to go over the code right away. It would be nice if there was someone > with more time and commit powers that could review some patches. > > > On Thu, Jan 31, 2013 at 11:23 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Ignored is not really a very positive sentiment.... >> >> We have a lot to deal with. So if there is a patch of this nature that is >> not committed yet, its just a matter of finding time to go over them. >> Please do not generate negative connotations. >> >> Its in tree now and I would appreciate some understanding in the future. >> >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/c41b6ae8/attachment.html From frederick at targointernet.com Thu Jan 31 19:51:37 2013 From: frederick at targointernet.com (Frederick Pruneau) Date: Thu, 31 Jan 2013 11:51:37 -0500 Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls In-Reply-To: References: <51092BBA.8030001@targointernet.com> <510A865C.505@targointernet.com> Message-ID: <510AA119.6080104@targointernet.com> Here is my log.: 2013-01-31 11:41:23.611280 [NOTICE] switch_channel.c:968 New Channel sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [0c742e6e-6bc5-11e2-93c2-2f40af36c49f] 2013-01-31 11:41:23.611280 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:23.611280 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:23.611280 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_NEW 2013-01-31 11:41:23.611280 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State NEW 2013-01-31 11:41:23.631279 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:23.631279 [DEBUG] sofia.c:1719 detaching session 0c742e6e-6bc5-11e2-93c2-2f40af36c49f 2013-01-31 11:41:25.531280 [NOTICE] switch_channel.c:968 New Channel sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [0d99b30e-6bc5-11e2-93c4-2f40af36c49f] 2013-01-31 11:41:25.531280 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:25.531280 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:25.531280 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_NEW 2013-01-31 11:41:25.531280 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State NEW 2013-01-31 11:41:25.551278 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:25.551278 [DEBUG] sofia.c:1719 detaching session 0d99b30e-6bc5-11e2-93c4-2f40af36c49f 2013-01-31 11:41:28.231280 [NOTICE] switch_channel.c:968 New Channel sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [0f340de0-6bc5-11e2-93c6-2f40af36c49f] 2013-01-31 11:41:28.231280 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:28.231280 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:28.231280 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_NEW 2013-01-31 11:41:28.231280 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State NEW 2013-01-31 11:41:28.251279 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:28.251279 [DEBUG] sofia.c:1719 detaching session 0f340de0-6bc5-11e2-93c6-2f40af36c49f 2013-01-31 11:41:33.631278 [WARNING] switch_core_state_machine.c:514 0c742e6e-6bc5-11e2-93c2-2f40af36c49f sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Abandoned 2013-01-31 11:41:33.631278 [DEBUG] switch_channel.c:2994 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Callstate Change DOWN -> HANGUP 2013-01-31 11:41:33.631278 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] 2013-01-31 11:41:33.631278 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [KILL] 2013-01-31 11:41:33.631278 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_HANGUP 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State HANGUP 2013-01-31 11:41:33.631278 [DEBUG] mod_sofia.c:503 Channel sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 hanging up, cause: WRONG_CALL_STATE 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:48 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Standard HANGUP, cause: WRONG_CALL_STATE 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State HANGUP going to sleep 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 11:41:33.631278 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_REPORTING 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State REPORTING 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:92 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Standard REPORTING, cause: WRONG_CALL_STATE 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State REPORTING going to sleep 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 11:41:33.631278 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:33.631278 [DEBUG] switch_core_session.c:1499 Session 116 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Locked, Waiting on external entities 2013-01-31 11:41:33.631278 [NOTICE] switch_core_session.c:1517 Session 116 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Ended 2013-01-31 11:41:33.631278 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [CS_DESTROY] 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Callstate Change HANGUP -> DOWN 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_DESTROY 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State DESTROY 2013-01-31 11:41:33.631278 [DEBUG] mod_sofia.c:396 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 SOFIA DESTROY 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:99 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Standard DESTROY 2013-01-31 11:41:33.631278 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State DESTROY going to sleep 2013-01-31 11:41:35.551279 [WARNING] switch_core_state_machine.c:514 0d99b30e-6bc5-11e2-93c4-2f40af36c49f sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Abandoned 2013-01-31 11:41:35.551279 [DEBUG] switch_channel.c:2994 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Callstate Change DOWN -> HANGUP 2013-01-31 11:41:35.551279 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] 2013-01-31 11:41:35.551279 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [KILL] 2013-01-31 11:41:35.551279 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_HANGUP 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State HANGUP 2013-01-31 11:41:35.551279 [DEBUG] mod_sofia.c:503 Channel sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 hanging up, cause: WRONG_CALL_STATE 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:48 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Standard HANGUP, cause: WRONG_CALL_STATE 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State HANGUP going to sleep 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 11:41:35.551279 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_REPORTING 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State REPORTING 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:92 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Standard REPORTING, cause: WRONG_CALL_STATE 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State REPORTING going to sleep 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 11:41:35.551279 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:35.551279 [DEBUG] switch_core_session.c:1499 Session 117 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Locked, Waiting on external entities 2013-01-31 11:41:35.551279 [NOTICE] switch_core_session.c:1517 Session 117 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Ended 2013-01-31 11:41:35.551279 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [CS_DESTROY] 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Callstate Change HANGUP -> DOWN 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_DESTROY 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State DESTROY 2013-01-31 11:41:35.551279 [DEBUG] mod_sofia.c:396 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 SOFIA DESTROY 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:99 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Standard DESTROY 2013-01-31 11:41:35.551279 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State DESTROY going to sleep 2013-01-31 11:41:38.231278 [WARNING] switch_core_state_machine.c:514 0f340de0-6bc5-11e2-93c6-2f40af36c49f sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Abandoned 2013-01-31 11:41:38.231278 [DEBUG] switch_channel.c:2994 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Callstate Change DOWN -> HANGUP 2013-01-31 11:41:38.231278 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] 2013-01-31 11:41:38.231278 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [KILL] 2013-01-31 11:41:38.231278 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_HANGUP 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State HANGUP 2013-01-31 11:41:38.231278 [DEBUG] mod_sofia.c:503 Channel sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 hanging up, cause: WRONG_CALL_STATE 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:48 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Standard HANGUP, cause: WRONG_CALL_STATE 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State HANGUP going to sleep 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State Change CS_HANGUP -> CS_REPORTING 2013-01-31 11:41:38.231278 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_REPORTING 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State REPORTING 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:92 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Standard REPORTING, cause: WRONG_CALL_STATE 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State REPORTING going to sleep 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State Change CS_REPORTING -> CS_DESTROY 2013-01-31 11:41:38.231278 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [BREAK] 2013-01-31 11:41:38.231278 [DEBUG] switch_core_session.c:1499 Session 118 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Locked, Waiting on external entities 2013-01-31 11:41:38.231278 [NOTICE] switch_core_session.c:1517 Session 118 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Ended 2013-01-31 11:41:38.231278 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 [CS_DESTROY] 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Callstate Change HANGUP -> DOWN 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) Running State Change CS_DESTROY 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State DESTROY 2013-01-31 11:41:38.231278 [DEBUG] mod_sofia.c:396 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 SOFIA DESTROY 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:99 sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060 Standard DESTROY 2013-01-31 11:41:38.231278 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/YYYYYYYYYY at XXX.XXX.XXX.XXX:5060) State DESTROY going to sleep Thanks! Fred Le 2013-01-31 11:15, Anthony Minessale a ?crit : > Its hard to tell without seeing a trace. You should turn up the > console logs and enabled debugging. > > sofia global siptrace on > console loglevel debug > > > > > On Thu, Jan 31, 2013 at 8:57 AM, Frederick Pruneau > > wrote: > > Hi Anthony, > > First of all, thanks for your reply! I appreciate your help! > > I checked conf files that could have changed since the new ip > address has been set. I tried to rollback as far as I can but did > not find the option that makes incoming calls not working. I > checked also my network settings. Everything seems to be good. I'm > probably missing something in freeswitch config files. > > Is there a file (or files) that I need to modify and put my new IP? > > Thanks again. > > Fred > > Le 2013-01-30 20:07, Anthony Minessale a ?crit : >> That means when the call was challenged for auth, it never >> replied with the new INVITE with www-auth header. >> You may have some network settings wrong making the followup >> message go to the wrong host. >> >> >> >> On Wed, Jan 30, 2013 at 8:18 AM, Frederick Pruneau >> > > wrote: >> >> Hi guys! >> >> I have a problem with my freeswitch server and I need some >> help. All >> incoming calls are not working since yesterday. All I did is >> to change >> the IP address. The only error message I have is this one: >> >> [WARNING] switch_core_state_machine.c:514 >> 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb >> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 >> Abandoned >> [NOTICE] switch_core_state_machine.c:517 Hangup >> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 >> [CS_NEW] >> [WRONG_CALL_STATE] >> >> What does Wrong call state mean and what can I do to fix this >> issue? >> >> Everything else is working. I can make outgoing calls and >> call other >> extensions. >> >> Thanks in advance. >> >> Fred >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/1937bb4b/attachment-0001.html From krice at freeswitch.org Thu Jan 31 19:51:41 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 31 Jan 2013 10:51:41 -0600 Subject: [Freeswitch-users] IAX In-Reply-To: <-1407881447721643425@unknownmsgid> Message-ID: Not going to happen any time soon. We had libiax support at one time but it doesn't support either trunking or registration... Whats wrong with SIP? The only real advantage to iax trunking is possible reduced bandwidth (nat is not a valid reason imo) and with RTP increasing the ptime to 40 ms along with using compressed codecs like G729 reduces bandwidth consumption to like 16kbps per call or about 5:1 over G711. (G711 at 20ms is about 80 kpbs and G729 at 20ms is about 24 kbps) K On 1/31/13 10:40 AM, "S. Scott" <8f27e956 at gmail.com> wrote: > I've done the google searches and it's not clear what the current > status of IAX facility is available in fs. We have a project needing > IAX trunk with registration support. > > With thanks, > > ????? > iThing: Big thumbs & little keys. Please excuse typo, spelling and > grammar errors ? Thought of the Day ? "With all this manure, there > must be a pony in here somewhere.? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From anthony.minessale at gmail.com Thu Jan 31 19:52:16 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 Jan 2013 10:52:16 -0600 Subject: [Freeswitch-users] IAX In-Reply-To: <-1407881447721643425@unknownmsgid> References: <-1407881447721643425@unknownmsgid> Message-ID: We do not support IAX. We did at one point but the code used was not maintained and became unstable and it was removed. Your best bet is to use an asterisk instance as an IAX to SIP conversion tool. Using their realtime thing, you could centralize the auth to the same source as your other FS based services. On Thu, Jan 31, 2013 at 10:40 AM, S. Scott <8f27e956 at gmail.com> wrote: > I've done the google searches and it's not clear what the current > status of IAX facility is available in fs. We have a project needing > IAX trunk with registration support. > > With thanks, > > ????? > iThing: Big thumbs & little keys. Please excuse typo, spelling and > grammar errors ? Thought of the Day ? "With all this manure, there > must be a pony in here somewhere.? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/bff09213/attachment.html From steveayre at gmail.com Thu Jan 31 19:54:16 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 31 Jan 2013 16:54:16 +0000 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: References: Message-ID: <8D9508D4-6825-4504-AF79-8FC90FA6A783@gmail.com> The developers are extremely busy being productive building the fantastic FreeSWITCH. It's understandable that the occasional minor ticket slips through the cracks. ;) If it's had no reply within a month or two sometimes a polite comment on the Jira ticket to bump it might be enough to catch the developers (such as Anthm's) attention. At the very least you can use it as an opportunity to check the patch applies against git head cleanly and update it if necessary as its more likely to get committed if its of good quality and needs no modifications. Steve On 31 Jan 2013, at 16:23, Anthony Minessale wrote: > Ignored is not really a very positive sentiment.... > > We have a lot to deal with. So if there is a patch of this nature that is not committed yet, its just a matter of finding time to go over them. Please do not generate negative connotations. > > Its in tree now and I would appreciate some understanding in the future. > > > > On Thu, Jan 31, 2013 at 9:26 AM, Andrew Cassidy wrote: >> Thanks, I'll give it a go later! >> >> On 31 January 2013 14:55, Vik Killa wrote: >>> >>>> Any not updated yet should be - if you find any such while testing raise a Jira on them (http://jira.freeswitch.org). >>> >>> ^^^ mod_nibblebill needs to be updated... >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> >> -- >> Andrew Cassidy BSc (Hons) MBCS SSCA >> Managing Director >> >> >> T 03300 100 960 F 03300 100 961 >> E andrew at cassidywebservices.co.uk >> W www.cassidywebservices.co.uk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/f39a0b82/attachment.html From bigx333 at gmail.com Thu Jan 31 19:56:22 2013 From: bigx333 at gmail.com (Nelson Luiz Ferraz de Camargo Penteado) Date: Thu, 31 Jan 2013 18:56:22 +0200 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: References: Message-ID: <76D13765-FDC2-4B13-8F4D-DB344CEA587E@gmail.com> This looks to be an issue to more related to the nibblebill module, I also have a patch for it in the JIRA for quite some months but patchs for the core seem to get responses rather quickly, I guess the maintainer of the nibblebill module is just inactive?. On Jan 31, 2013, at 6:48 PM, Anthony Minessale wrote: > Try to shift your mentality so you try to get patches to make it into releases more importantly than into HEAD. If the patch still compiles then its not a big deal if its not in yet as long as it cleanly applies and makes it in before a tag. > > We are separating the release structure to build multiple channels. This is not easy. If we don't do it people will get mad that we have no stable channel. If we do it, people get mad we are too slow to add their patches. Unfortunately someone will always be mad. Even George Lucas gets yelled at for not making Star Wars right. Sadly I must make decisions that are not 100% favorable but your patch is a welcomed addition since we clearly missed it when we changed all the other modules. Going forward think of it terms of seasons. We will have release season where we slow down and make sure the code is stable and we will have development season when we change everything. 1.5 is coming out soon as the HEAD for 1.4. Then we can look at some more patches that have been lingering. > > > > > > On Thu, Jan 31, 2013 at 10:35 AM, Vik Killa wrote: > I apologize for my negative tone. It's just a bit frustrating having patches sit there for months. I understand you are busy and I don't expect you to go over the code right away. It would be nice if there was someone with more time and commit powers that could review some patches. > > > On Thu, Jan 31, 2013 at 11:23 AM, Anthony Minessale wrote: > Ignored is not really a very positive sentiment.... > > We have a lot to deal with. So if there is a patch of this nature that is not committed yet, its just a matter of finding time to go over them. Please do not generate negative connotations. > > Its in tree now and I would appreciate some understanding in the future. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/06326acd/attachment-0001.html From frederick at targointernet.com Thu Jan 31 20:09:39 2013 From: frederick at targointernet.com (Frederick Pruneau) Date: Thu, 31 Jan 2013 12:09:39 -0500 Subject: [Freeswitch-users] [CS_NEW] [WRONG_CALL_STATE] with incoming calls In-Reply-To: <06E0038D-469F-47F1-8449-C6B590AAECA0@gmail.com> References: <51092BBA.8030001@targointernet.com> <510A865C.505@targointernet.com> <06E0038D-469F-47F1-8449-C6B590AAECA0@gmail.com> Message-ID: <510AA553.9060105@targointernet.com> I changed sip-ip and rtp-ip settings for my ip address. I restarted freeswitch. The problem is still there. My FS server is not behind nat. Thanks for your help! Fred Le 2013-01-31 11:46, Steven Ayre a ?crit : > Check the sip-ip rtp-ip settings on your Sofia profiles. They may be > using a variable such as local_ip4, if so check that too. > > Did you restart FS or at least the relevant profiles after changing > the IP? > > Note that if you're not setting it explicitly it may be autodetecting > your IP, in which case that might not happen correctly if your server > now has multiple IPs. Or using the old one if FS wasn't restarted. > > Also remember Sofia cannot listen on 0.0.0.0 > > If your FS is behind NAT then that also may imply a few other settings > need checking too (ext-sip-ip etc). > > Steve > > > > On 31 Jan 2013, at 14:57, Frederick Pruneau > > wrote: > >> Hi Anthony, >> >> First of all, thanks for your reply! I appreciate your help! >> >> I checked conf files that could have changed since the new ip address >> has been set. I tried to rollback as far as I can but did not find >> the option that makes incoming calls not working. I checked also my >> network settings. Everything seems to be good. I'm probably missing >> something in freeswitch config files. >> >> Is there a file (or files) that I need to modify and put my new IP? >> >> Thanks again. >> >> Fred >> >> Le 2013-01-30 20:07, Anthony Minessale a ?crit : >>> That means when the call was challenged for auth, it never replied >>> with the new INVITE with www-auth header. >>> You may have some network settings wrong making the followup message >>> go to the wrong host. >>> >>> >>> >>> On Wed, Jan 30, 2013 at 8:18 AM, Frederick Pruneau >>> > >>> wrote: >>> >>> Hi guys! >>> >>> I have a problem with my freeswitch server and I need some help. All >>> incoming calls are not working since yesterday. All I did is to >>> change >>> the IP address. The only error message I have is this one: >>> >>> [WARNING] switch_core_state_machine.c:514 >>> 7b24f8fa-6ae6-11e2-a6be-25229cd2cfbb >>> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 Abandoned >>> [NOTICE] switch_core_state_machine.c:517 Hangup >>> sofia/internal/XXX at XXX.XXX.XXX.XXX:5060 [CS_NEW] [WRONG_CALL_STATE] >>> >>> What does Wrong call state mean and what can I do to fix this issue? >>> >>> Everything else is working. I can make outgoing calls and call other >>> extensions. >>> >>> Thanks in advance. >>> >>> Fred >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> >>> googletalk:conf+888 at conference.freeswitch.org >>> >>> pstn:+19193869900 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/981def50/attachment.html From a.venugopan at mundio.com Thu Jan 31 20:21:32 2013 From: a.venugopan at mundio.com (Archana Venugopan) Date: Thu, 31 Jan 2013 17:21:32 +0000 Subject: [Freeswitch-users] ICTFAX installation in FS Message-ID: <592A9CF93E12394E8472A6CC66E66BF23559FE@Mail-Kilo.squay.com> Hi, Am trying to install ICTFAX. As per this site ictfax-20-installation-guide I followed the instructions. But got stuck in the below procedures 1. Create a symbolic link for /usr/ictfax in /var/www/html ln -s /usr/ictfax /var/www/html/ictfax ln: creating symbolic link `/var/www/html/ictfax': No such file or directory. There is no ictfax directory in the folder. So what to do? 2. Now visit http://DOMAIN.COM/ictfax and follow the installation instructions for ICTFax (drupal based) front end installation When I click on the link its giving 404 error. Can anyone please guide me on how to proceed further on this. Many thanks Regards, Archana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/d805e082/attachment.html From bdfoster at endigotech.com Thu Jan 31 20:31:15 2013 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 31 Jan 2013 12:31:15 -0500 Subject: [Freeswitch-users] Postgres in the Core In-Reply-To: <76D13765-FDC2-4B13-8F4D-DB344CEA587E@gmail.com> References: <76D13765-FDC2-4B13-8F4D-DB344CEA587E@gmail.com> Message-ID: <5C4B5BDA-49F7-4485-ACA5-46E614EB6E00@endigotech.com> Yea I'd imagine he is, he runs 2600hz... Sent from my iPhone On Jan 31, 2013, at 11:56 AM, Nelson Luiz Ferraz de Camargo Penteado wrote: > This looks to be an issue to more related to the nibblebill module, I also have a patch for it in the JIRA for quite some months but patchs for the core seem to get responses rather quickly, I guess the maintainer of the nibblebill module is just inactive?. > > On Jan 31, 2013, at 6:48 PM, Anthony Minessale wrote: > >> Try to shift your mentality so you try to get patches to make it into releases more importantly than into HEAD. If the patch still compiles then its not a big deal if its not in yet as long as it cleanly applies and makes it in before a tag. >> >> We are separating the release structure to build multiple channels. This is not easy. If we don't do it people will get mad that we have no stable channel. If we do it, people get mad we are too slow to add their patches. Unfortunately someone will always be mad. Even George Lucas gets yelled at for not making Star Wars right. Sadly I must make decisions that are not 100% favorable but your patch is a welcomed addition since we clearly missed it when we changed all the other modules. Going forward think of it terms of seasons. We will have release season where we slow down and make sure the code is stable and we will have development season when we change everything. 1.5 is coming out soon as the HEAD for 1.4. Then we can look at some more patches that have been lingering. >> >> >> >> >> >> On Thu, Jan 31, 2013 at 10:35 AM, Vik Killa wrote: >>> I apologize for my negative tone. It's just a bit frustrating having patches sit there for months. I understand you are busy and I don't expect you to go over the code right away. It would be nice if there was someone with more time and commit powers that could review some patches. >>> >>> >>> On Thu, Jan 31, 2013 at 11:23 AM, Anthony Minessale wrote: >>>> Ignored is not really a very positive sentiment.... >>>> >>>> We have a lot to deal with. So if there is a patch of this nature that is not committed yet, its just a matter of finding time to go over them. Please do not generate negative connotations. >>>> >>>> Its in tree now and I would appreciate some understanding in the future. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/81c1a1ef/attachment-0001.html From Hector.Geraldino at ipsoft.com Thu Jan 31 22:18:41 2013 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Thu, 31 Jan 2013 19:18:41 +0000 Subject: [Freeswitch-users] Execute app in sync mode when using event sockets Message-ID: Hi, I have a java app that uses mod_event_socket (inbound mode) to control FreeSWITCH. I can send api commands which, by default, are synchronous, so the thread locks until the command is successfully executed. The problem I'm facing is when I have to execute commands that are in other modules (like playback, or speak). As they are not api commands, I must send an execute appName + arguments, which by nature is executed in async mode. What I want to achieve is to execute an speak command and wait until the TTS operation ends (blocking the thread). This sounds like a trivial use case: play some audio (TTS, file) and then execute the rest of the logic. It is easy to accomplish on the dialplan, but I just can't figure out how to do it on an ESL application. Here's what I've tried so far: + send execute speak command + api sleep and wait for the CHANNEL_EXECUTE_COMPLETE (to simulate a lock [sleep] and continue [break after event is received]), but the event arrives after the sleep command completes. + use uuid_broadcast uuid speak::args, doesn't work Any hints, ideas are welcomed. Thank you! Hector -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/6a579ccb/attachment.html From krice at freeswitch.org Thu Jan 31 22:27:41 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 31 Jan 2013 13:27:41 -0600 Subject: [Freeswitch-users] ZRTP Cache file location on FreeSWITCH fixed/updated Message-ID: Hey Guys, If you have noticed theres been an issue with ZRTP enabled FreeSWITCH where ZRTP keeps a cache of hashs identifying previously contacted ZRTP endpoints. (This is part of the mechanism to help detect Man in the Middle attacks). Prior to today that cache file was being improperly named ! And was being dropped in the Directory FreeSWITCH was being ran from. Travis Cross and I got a patch together today and it has been pushed into tree. Starting today the file has been moved over to the proper FreeSWITCH db directory (/usr/local/freeswitch/db on default from source installs) with the name zrtp.dat . If you are currently using this functionality, you can keep your cache around by moving it to the new location. (ex: # cd /usr/local/freeswitch && mv \! db/zrtp.dat ) This patch is currently only in master, but will be rolled into the pending stable updates. Thanks for the help Travis! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/bb4fee89/attachment.html From sdevoy at bizfocused.com Thu Jan 31 23:13:20 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 31 Jan 2013 15:13:20 -0500 Subject: [Freeswitch-users] simple install question Message-ID: <091801cdffef$6a4229c0$3ec67d40$@bizfocused.com> Hi, I want to switch from the stable release version to the git head and rebuild in hopes of ridding myself of the delay in connection of the audio. On almost all my calls I must hesitate about 1 second before saying hello or the caller does not hear me. I have seen many discussions saying that this is probably fixed in the git root. So here are my questions: The procedure appears to be: Log in to root cd /usr/local/src/freeswitch git checkout master Which responds: Branch master set up to track remote branch master from origin. Switched to a new branch 'master' Then just: make current Is that correct? Do I need to stop the running FS while running this? If so, how long will I be down? (ball park it) I realize this is a RTFM kind of question, but I can't risk an outage. I need confirmation from someone who understands. Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/57f45769/attachment.html From avi at avimarcus.net Thu Jan 31 23:24:29 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 31 Jan 2013 22:24:29 +0200 Subject: [Freeswitch-users] simple install question In-Reply-To: <091801cdffef$6a4229c0$3ec67d40$@bizfocused.com> References: <091801cdffef$6a4229c0$3ec67d40$@bizfocused.com> Message-ID: Follow the instructions at: http://wiki.freeswitch.org/wiki/Installation_Guide#Reverting_to_an_Earlier_Commit_in_Git to fully clean out the old build files. -Avi On Thu, Jan 31, 2013 at 10:13 PM, Sean Devoy wrote: > Hi,**** > > ** ** > > I want to switch from the stable release version to the git head and > rebuild in hopes of ridding myself of the delay in connection of the > audio. On almost all my calls I must hesitate about 1 second before saying > hello or the caller does not hear me. I have seen many discussions saying > that this is probably fixed in the git root. So here are my questions:*** > * > > ** ** > > The procedure appears to be:**** > > Log in to root**** > > cd /usr/local/src/freeswitch**** > > git checkout master**** > > Which responds: Branch master set up to track remote branch master from > origin.**** > > Switched to a new branch 'master'**** > > Then just: **** > > make current**** > > ** ** > > Is that correct?**** > > Do I need to stop the running FS while running this?**** > > If so, how long will I be down? (ball park it)**** > > ** ** > > I realize this is a RTFM kind of question, but I can?t risk an outage. I > need confirmation from someone who understands.**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/e7c4e6a4/attachment.html From jnvines at gmail.com Thu Jan 31 23:24:59 2013 From: jnvines at gmail.com (Nick Vines) Date: Thu, 31 Jan 2013 12:24:59 -0800 Subject: [Freeswitch-users] simple install question In-Reply-To: <091801cdffef$6a4229c0$3ec67d40$@bizfocused.com> References: <091801cdffef$6a4229c0$3ec67d40$@bizfocused.com> Message-ID: Stolen from an email from Ken. "If you need to switch between branches, you _must_ git clean -fdx and rm all the existing binaries in your existing FreeSWITCH install for the mod lib and bin directories or you will most likely end up with build skew and crazy weird segfaults..." I think it is possible to install while freeswitch is running, but highly advised against. Just do it at a 3am maintenance outage. You should be able to run the bootstrap and configure before you shutdown your FS server to reduce downtime. On the base rackspace cloud instance (512mb version), it takes about 10 min total from shutdown to startup of FS. On Thu, Jan 31, 2013 at 12:13 PM, Sean Devoy wrote: > Hi,**** > > ** ** > > I want to switch from the stable release version to the git head and > rebuild in hopes of ridding myself of the delay in connection of the > audio. On almost all my calls I must hesitate about 1 second before saying > hello or the caller does not hear me. I have seen many discussions saying > that this is probably fixed in the git root. So here are my questions:*** > * > > ** ** > > The procedure appears to be:**** > > Log in to root**** > > cd /usr/local/src/freeswitch**** > > git checkout master**** > > Which responds: Branch master set up to track remote branch master from > origin.**** > > Switched to a new branch 'master'**** > > Then just: **** > > make current**** > > ** ** > > Is that correct?**** > > Do I need to stop the running FS while running this?**** > > If so, how long will I be down? (ball park it)**** > > ** ** > > I realize this is a RTFM kind of question, but I can?t risk an outage. I > need confirmation from someone who understands.**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/2cfe6fcf/attachment-0001.html From avi at avimarcus.net Thu Jan 31 23:31:50 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 31 Jan 2013 22:31:50 +0200 Subject: [Freeswitch-users] simple install question In-Reply-To: References: <091801cdffef$6a4229c0$3ec67d40$@bizfocused.com> Message-ID: It would be nice to have instructions for this somewhere... some people tar up their old build directory, or at least the compiled files. That way if the build/install goes bonkers, they can untar and re-install their old binaries. -Avi On Thu, Jan 31, 2013 at 10:24 PM, Nick Vines wrote: > Stolen from an email from Ken. > > "If you need to switch between branches, you _must_ git clean -fdx and rm > all > the existing binaries in your existing FreeSWITCH install for the mod lib > and bin directories or you will most likely end up with build skew and > crazy > weird segfaults..." > > I think it is possible to install while freeswitch is running, but highly > advised against. Just do it at a 3am maintenance outage. You should be > able to run the bootstrap and configure before you shutdown your FS server > to reduce downtime. > > On the base rackspace cloud instance (512mb version), it takes about 10 > min total from shutdown to startup of FS. > > > On Thu, Jan 31, 2013 at 12:13 PM, Sean Devoy wrote: > >> Hi,**** >> >> ** ** >> >> I want to switch from the stable release version to the git head and >> rebuild in hopes of ridding myself of the delay in connection of the >> audio. On almost all my calls I must hesitate about 1 second before saying >> hello or the caller does not hear me. I have seen many discussions saying >> that this is probably fixed in the git root. So here are my questions:** >> ** >> >> ** ** >> >> The procedure appears to be:**** >> >> Log in to root**** >> >> cd /usr/local/src/freeswitch**** >> >> git checkout master**** >> >> Which responds: Branch master set up to track remote branch master from >> origin.**** >> >> Switched to a new branch 'master'**** >> >> Then just: **** >> >> make current**** >> >> ** ** >> >> Is that correct?**** >> >> Do I need to stop the running FS while running this?**** >> >> If so, how long will I be down? (ball park it)**** >> >> ** ** >> >> I realize this is a RTFM kind of question, but I can?t risk an outage. I >> need confirmation from someone who understands.**** >> >> ** ** >> >> Thanks,**** >> >> Sean**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130131/7fc79890/attachment.html From krice at freeswitch.org Thu Jan 31 23:47:56 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 31 Jan 2013 14:47:56 -0600 Subject: [Freeswitch-users] simple install question In-Reply-To: Message-ID: The directions you referenced in a previous email are the same reguardless of changing between branches or going to older hashes... Sometimes you can get away with it, but it will eventually cause issues... If you are installing to /usr/local/freeswitch , you can easily tar zcvf freeswitch.DATE.tgz freeswitch (from /usr/local/freeswitch) of course... K On 1/31/13 2:31 PM, "Avi Marcus" wrote: > It would be nice to have instructions for this somewhere... > some people tar up their old build directory, or at least the compiled files. > That way if the build/install goes bonkers, they can untar and re-install > their old binaries. > > -Avi > > On Thu, Jan 31, 2013 at 10:24 PM, Nick Vines wrote: >> Stolen from an email from Ken.? >> >> "If you need to switch between branches, you _must_ git clean -fdx?and rm all >> the existing binaries in your existing FreeSWITCH install for the mod lib >> and bin directories or you will most likely end up with build skew and crazy >> weird segfaults..." >> >> I think it is possible to install while freeswitch is running, but highly >> advised against. Just do it at a 3am maintenance outage.?You should be able >> to run the bootstrap and configure before you shutdown your FS server to >> reduce downtime.? >> >> On the base rackspace cloud instance (512mb version), it takes about 10 min >> total from shutdown to startup of FS.? >> >> >> On Thu, Jan 31, 2013 at 12:13 PM, Sean Devoy wrote: >>> Hi, >>> ? >>> I want to switch from the stable release version to the git head and rebuild >>> in hopes of ridding myself of the delay in connection of the audio.? On >>> almost all my calls I must hesitate about 1 second before saying hello or >>> the caller does not hear me.? I have seen many discussions saying that this >>> is probably fixed in the git root.? So here are my questions: >>> ? >>> The procedure appears to be: >>> Log in to root >>> cd /usr/local/src/freeswitch >>> git checkout master >>> Which responds: Branch master set up to track remote branch master from >>> origin. >>> ????????????????????????????????? Switched to a new branch 'master' >>> Then just: >>> make current >>> ? >>> Is that correct? >>> Do I need to stop the running FS while running this? >>> If so, how long will I be down?? (ball park it) >>> ? >>> I realize this is a RTFM kind of question, but I can?t risk an outage.? I >>> need confirmation from someone who understands. >>> ? >>> Thanks, >>> Sean >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... 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