July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem with pause
by Fabio Bigliardi 14 Jul '11

14 Jul '11

Hi all, I'm trying to have an audio stream played back on a channel and to pause/resume it through API commands on event socket. Below you can find the sequence of commands and outputs: $ telnet 127.0.0.1 8021 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Content-Type: auth/request auth ClueCon Content-Type: command/reply Reply-Text: +OK accepte api originate user/1003 &playback(${sounds_dir}/jazz.wav) Content-Type: api/response Content-Length: 41 +OK 8fcdb494-82a4-42c4-b011-ecca1da0ade2 api pause 8fcdb494-82a4-42c4-b011-ecca1da0ade2 on Content-Type: api/response Content-Length: 14 *-ERR no reply* ** * * The "pause" command is successful (the stream is really paused) but the api response indicates an error. Same output with the CLI. Is it correct? Thank you very much. Fabio Bigliardi

2 1

Cisco 7960 registration issue using mod_skinny
by Manjiri Deshpande 14 Jul '11

14 Jul '11

Hi, From last week I was trying to register Cisco phone with FreeSwitch using mod_skinny.I am using solarwinds as TFTP server . But registration is not successful. Phone cycles through the process of TFTP download and sending register message along with some more SKINNY messages to FreeSwitch. Also on FreeSwitch Console I get message as "UCM-Closed-TCP" Attached are the wireshark capture for phone and FreeSwitch.log file From wireshark capture we can clearly see that phone sends "RegisterMessage" SKINNY message in a cycle. This is the consistent flow. Also attaching SEP[MAC].cnf.xml file which I am using. Ph IP Addr : 192.1681.1.205 FreeSwitchIP Addr : 192.168.1.116 Please have a look at this issue and let me know if I am missing anything here Thanks, Manjiri

2 1

session.recordFile create empy wav in while loop
by Daniel Knoll 14 Jul '11

14 Jul '11

Hey Guys, i have a strange Problem with session.recordFile in a while loop and i don't know how to solve it. The first run in the loop is absolutly ok, session.recordFile is creating a good clean wavefile. but in the second, third .. run, every wav file has a size of 47724 byte and no audio content. Please help me, because it is an urgent case. Thankx for getting help. Daniel

4 10

FreeTDM - Sangoma B700 - ISDN connection questions - UK
by John 14 Jul '11

14 Jul '11

Hello, I have just set up a FreeSWITCH box with a Sangoma B700 connected to 2 x ISDN2e lines (each 2B+D, so 4 voice channels) in UK, so supplied by BT Openreach. There are a number of anomalies that I am trying to solve. [One of these is that 1 line is dead, but that is for BT Openreach to resolve rather than anyone on this list.] I have a few questions, but as they are all related, I hope you don't mind them in one post. Some basics first. The box is an Atom dual-core with 2GB of memory and a Sangoma B700 card. It's built with Ubuntu 10.04.2 LTS server 64-bit, patched and up to date, and also runs dhcpd, lighttpd and sshd. I built the Sangoma ISDN libraries and FreeSWITCH using the latest git versions I could ('make current' about 2 weeks ago). We have incoming and outgoing calls working, but some incoming calls ring in the caller's ear, but nothing appears on the FreeSWITCH console, and others fail. Rebooting the server cures this. We have had problems with lines being disconnected and then reconnected, and it seems that FreeTDM/wanrouter/whatever doesn't recognise previously disconnected lines coming back into use, because a reboot finds lines that were previously reported disconnected ('wanrouter status' or 'ftdm list'). We have ongoing problems with one line that is 'disconnected' (wanpipe2/wp2) but the queries I am describing below apply equally when both lines are connected and working. 1. There seems to be a lot of ISDN 'chatter' with channels going up and down all the time even when the switch is completely idle. Is this normal? Here is my /log 7 showing two of the cycles (which appear to be about 50s apart): 2011-06-06 22:01:54.992467 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681 [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect initiated(263) 2011-06-06 22:02:29.952464 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:748 [SNGISDN Q931] s1: Interface: Down(261): Dchan(285) 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 [s1c1][1:1] Signalling link status changed to DOWN 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 [s1c2][1:2] Signalling link status changed to DOWN 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 [s1c3][1:3] Signalling link status changed to DOWN 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200 [s1c1][1:1] Setting availability rate to:5 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel sig [SIGSTATUS_CHANGED] 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186 [s1c1][1:1] Setting availability rate to:5 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200 [s1c2][1:2] Setting availability rate to:5 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:1 signalling changed to :DOWN 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186 [s1c2][1:2] Setting availability rate to:5 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel sig [SIGSTATUS_CHANGED] 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:200 [s1c3][1:3] Setting availability rate to:5 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:2 signalling changed to :DOWN 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_support.c:186 [s1c3][1:3] Setting availability rate to:5 2011-06-06 22:02:29.952464 [DEBUG] mod_freetdm.c:2268 got clear channel sig [SIGSTATUS_CHANGED] 2011-06-06 22:02:29.952464 [INFO] mod_freetdm.c:2345 1:3 signalling changed to :DOWN 2011-06-06 22:02:29.952464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631 Received RESTART CFM (dChan:1 ces:0 type:1) 2011-06-06 22:02:29.952464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179 Receved RESTART, but Restart Indicator IE not present 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681 [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with F-bit = 1(258) 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631 Received RESTART CFM (dChan:1 ces:0 type:0) 2011-06-06 22:02:29.972464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729 [SNGISDN Q931] s1: Interface: UP(260): Dchan(285) 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 [s1c1][1:1] Signalling link status changed to UP 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 [s1c2][1:2] Signalling link status changed to UP 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 [s1c3][1:3] Signalling link status changed to UP 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200 [s1c1][1:1] Setting availability rate to:10 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186 [s1c1][1:1] Setting availability rate to:10 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200 [s1c2][1:2] Setting availability rate to:10 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186 [s1c2][1:2] Setting availability rate to:10 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:200 [s1c3][1:3] Setting availability rate to:10 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_support.c:186 [s1c3][1:3] Setting availability rate to:10 2011-06-06 22:02:29.972464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179 Receved RESTART, but Restart Indicator IE not present 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel sig [SIGSTATUS_CHANGED] 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:1 signalling changed to :UP 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel sig [SIGSTATUS_CHANGED] 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:2 signalling changed to :UP 2011-06-06 22:02:29.972464 [DEBUG] mod_freetdm.c:2268 got clear channel sig [SIGSTATUS_CHANGED] 2011-06-06 22:02:29.972464 [INFO] mod_freetdm.c:2345 1:3 signalling changed to :UP 2011-06-06 22:03:19.912462 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681 [SNGISDN Q921] wp1: Protocol: Data Link connection DN(4): Disconnect initiated(263) 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:681 [SNGISDN Q921] wp1: Protocol: Data Link connection UP(3): UA frame with F-bit = 1(258) 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:631 Received RESTART CFM (dChan:1 ces:0 type:0) 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:1179 Receved RESTART, but Restart Indicator IE not present 2011-06-06 22:03:19.932483 [INFO] ftmod_sangoma_isdn_stack_rcv.c:729 [SNGISDN Q931] s1: Interface: UP(260): Dchan(285) 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 [s1c1][1:1] Signalling link status changed to UP 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 [s1c2][1:2] Signalling link status changed to UP 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_cntrl.c:43 [s1c3][1:3] Signalling link status changed to UP 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200 [s1c1][1:1] Setting availability rate to:10 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186 [s1c1][1:1] Setting availability rate to:10 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200 [s1c2][1:2] Setting availability rate to:10 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186 [s1c2][1:2] Setting availability rate to:10 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:200 [s1c3][1:3] Setting availability rate to:10 2011-06-06 22:03:19.932483 [DEBUG] ftmod_sangoma_isdn_support.c:186 [s1c3][1:3] Setting availability rate to:10 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel sig [SIGSTATUS_CHANGED] 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:1 signalling changed to :UP 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel sig [SIGSTATUS_CHANGED] 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:2 signalling changed to :UP 2011-06-06 22:03:19.932483 [DEBUG] mod_freetdm.c:2268 got clear channel sig [SIGSTATUS_CHANGED] 2011-06-06 22:03:19.932483 [INFO] mod_freetdm.c:2345 1:3 signalling changed to :UP freeswitch@internal> ftdm list +OK span: 1 (wp1) type: Sangoma (ISDN) physical_status: ok signaling_status: UP chan_count: 3 dialplan: XML context: public dial_regex: fail_dial_regex: hold_music: analog_options: none +OK span: 2 (wp2) type: Sangoma (ISDN) physical_status: alarmed signaling_status: DOWN chan_count: 3 dialplan: XML context: public dial_regex: fail_dial_regex: hold_music: analog_options: none +OK span: 3 (FXS) type: analog physical_status: ok signaling_status: UP chan_count: 2 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options: none # wanrouter status Devices currently active: wanpipe1 wanpipe2 wanpipe3 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1 | N/A | A500/B700| 20 | 0 | 1 | N/A | 0 | wanpipe2 | N/A | A500/B700| 20 | 0 | 1 | N/A | 0 | wanpipe3 | N/A | A200/A400/B600/B700/B800| 20 | 0 | 1 | N/A | 0 | Wanrouter Status: Device name | Protocol | Station | Status | wanpipe1 | AFT ISDN | N/A | Connected | wanpipe2 | AFT ISDN | N/A | Disconnected | wanpipe3 | A-ANALOG | N/A | Connected | # cat /etc/wanpipe/wanrouter.rc #!/bin/sh # .. comments snipped ... ROUTER_BOOT=YES WAN_CONF_DIR=/etc/wanpipe WAN_INTR_DIR=/etc/wanpipe/interfaces WAN_BIN_DIR=/usr/sbin WAN_LOG=/var/log/wanrouter WAN_LOCK=/var/lock/wanrouter WAN_LOCK_DIR=/var/lock WAN_IP_FORWARD=NO NEW_IF_TYPE=NO WAN_LIB_DIR=/etc/wanpipe/lib WAN_ADSL_LIST=/etc/wanpipe/wan_adsl.list WAN_ANNEXG_LOAD=NO WAN_SCTP_LOAD=NO WAN_LIP_LOAD=NO WAN_DYN_WANCONFIG=NO WAN_SCRIPTS_DIR=/etc/wanpipe/scripts WAN_FIRMWARE_DIR=/etc/wanpipe/firmware WAN_DEVICES_REV_STOP_ORDER=YES WAN_DEVICES="wanpipe1 wanpipe2 wanpipe3 " # cat /etc/wanpipe/wanpipe1.conf #================================================ # WANPIPE1 Configuration File #================================================ # # Note: This file was generated automatically # by /usr/local/sbin/setup-sangoma program. # # If you want to edit this file, it is # recommended that you use wancfg program # to do so. #================================================ # Sangoma Technologies Inc. #================================================ [devices] wanpipe1 = WAN_AFT_ISDN_BRI, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 0 PCIBUS = 5 FE_MEDIA = BRI FE_LINE = 1 TDMV_LAW = ALAW RM_BRI_CLOCK_MASTER = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS Blue Alarm and keep line down #wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode #wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode TDMV_HW_DTMF = YES # YES: receive dtmf events from hardware TDMV_HW_FAX_DETECT = YES # YES: receive fax 1100hz events from hardware HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation enabled with nlp (default) # OCT_SPEECH: improves software tone detection by disabling NLP (echo possible) # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions. HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of incoming media (must have hwdtmf enabled) HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the line - could break fax HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo cancelation HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software tone detection (possible echo) HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal [w1g1] ACTIVE_CH = ALL TDMV_HWEC = YES MTU = 80 ... and wanpipe2 was also automatically generated so just has '2' for FE_LINE and TDMV_SPAN instead of '1'. # cat /usr/local/freeswitch/conf/freetdm.conf [span wanpipe wp1] trunk_type => bri group=1 b-channel => 1:1-2 d-channel => 1:3 [span wanpipe wp2] trunk_type => bri group=1 b-channel => 2:1-2 d-channel => 2:3 [span wanpipe FXS] name => freetdm trunk_type => fxs group => grp2 fxs-channel => 3:1 trunk_type => fxs group => grp2 fxs-channel => 3:2 # cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml <configuration name="freetdm.conf" description="Freetdm Configuration"> <settings> <param name="debug" value="0"/>    </settings> <config_profiles> <profile name="my_bri_te_1"> <param name="switchtype" value="euroisdn" /> <param name="interface" value="cpe"/> <param name="facility" value="no" /> </profile> </config_profiles> <sangoma_bri_spans> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </sangoma_bri_spans> <analog_spans>  <param name="dialplan" value="XML"/> <param name="context" value="default"/>     </analog_spans> </configuration> What other config files do I need to supply? I don't want to deluge the list any more in my first post! 2. We have 2 Single Number DDI numbers configured over the 4 channels. I want to set up outgoing calls so that they can appear to come from either of the two numbers, but at present the outgoing CLI appears to be overridden by the telco (BT) to only use one of the numbers. Has anyone got this working in UK, and what is the format for the outbound_caller_id_number: last 6 digits or full 11 digits? I note that the inbound called number is only the last 6 digits. 3. I have built FreeSWITCH from git and installed at /usr/local/... and then followed the steps on the Ubuntu page in the Wiki to set up the run control scripts, etc, and run FS non-root as freeswitch:daemon. With FreeTDM, I have discovered that the /dev/wan* devices are owned by root:root, and so are inaccessible to FS running as non-root. So for now I have added a line in /etc/init.d/freeswitch to 'chgrp freeswitch /dev/wan*'. This is not the most elegant solution, because 'wanrouter restart' (which seems to be my best friend at the moment) resets the ownership to root:root. I have tried grepping to see where the mknods are for these devices, but have been unsuccessful. Is there a better place to 'permanently' change the device ownership? Thanks for all the great support I have already got just from editing my Wiki User page; this is a friendly group! John

13 31

Javascript from javascript == session freeze
by Boris Kovalenko 14 Jul '11

14 Jul '11

Hello! I modified my DISA javascript to use 'originate' back to my FS with collected number. The context where the call is placed calls another simple javascript application. And when application is called my session freezes. If I remove call to javascript application all is ok. So am I missed something in docs and I can not call this way or should I open JIRA? -- Regards, Boris

1 0

How to run flex client correctly?
by Dmitry Kravchenko 14 Jul '11

14 Jul '11

Hi! How to run flex client correctly? If I just put a repository files from here http://fisheye.freeswitch.org/browse/freeswitch.git/clients/flex into webserver root, and open freeswitch.html, I get damaged web page with apparently no design. Looks like stylesheets not work. No question about allowing microphone appears. If I open swf file in flashplayer, it is just blank. Changing flashvars to correct freeswitch installation URL does not help. Running mxmlc does not help (although it runs without errors). How correctly working client should look like? Where to put freeswitch credentials? Thanks.

1 0

Call drops after 30 seconds
by Matthew Ralston 14 Jul '11

14 Jul '11

Hi, I'm having a problem at the moment with calls being successfully set up, with two-way audio, being terminated by FreeSWITCH after 30 seconds. Internal calls (i.e. between SIP phones on the same LAN segment as the FreeSWITCH box) work flawlessly. The problem arises when at least one of the handsets is located elsewhere on the Internet. This behaviour is exhibited under the following circumstances: - A-leg only call, e.g. to voicemail when the handset is at another location on the Internet - A-leg-B-leg call if one or both of the handsets are at another location on the Internet - Inbound calls from our external SIP provider - Outbound calls to our external SIP provider So it is obvious that the problem is related to the SIP going via the Internet, but I'm having trouble understanding why. Whilst debugging this problem I have placed the FreeSWITCH box is in the DMZ on our router, so there should not be any ports blocked. The FreeSWITCH box itself is not running a software firewall. The calls themselves are absolutely fine for the first 30 seconds - each party can hear the other talking fine. The fact that the call is consistently dropped after 30 seconds (give or take a second or two for PDD) suggests that some timeout is being triggered. When FreeSWITCH terminates the call, the following is logged to the console: 2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel sofia/internal/1006(a)public.ip.removed entering state [terminating][0] 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641 (sofia/internal/1006(a)public.ip.removed) Callstate Change ACTIVE -> HANGUP 2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup sofia/internal/1006(a)public.ip.removed [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657 Send signal sofia/internal/1006(a)public.ip.removed [KILL] 2011-06-22 13:33:50.514941 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006(a)public.ip.removed [BREAK] 2011-06-22 13:33:50.534966 [DEBUG] switch_ivr_play_say.c:1649 done playing file 2011-06-22 13:33:50.625988 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-press.wav] (en:en) 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:2063 sofia/internal/1006(a)public.ip.removed skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1006(a)public.ip.removed) State EXECUTE going to sleep 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1006(a)public.ip.removed) Running State Change CS_HANGUP 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1006(a)public.ip.removed) State HANGUP 2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458 Channel sofia/internal/1006(a)public.ip.removed hanging up, cause: NORMAL_UNSPECIFIED 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1006(a)public.ip.removed Standard HANGUP, cause: NORMAL_UNSPECIFIED 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1006(a)public.ip.removed) State HANGUP going to sleep 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1006(a)public.ip.removed) State Change CS_HANGUP -> CS_REPORTING 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006(a)public.ip.removed [BREAK] 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1006(a)public.ip.removed) Running State Change CS_REPORTING 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1006(a)public.ip.removed) State REPORTING 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1006(a)public.ip.removed Standard REPORTING, cause: NORMAL_UNSPECIFIED 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1006(a)public.ip.removed) State REPORTING going to sleep 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1006(a)public.ip.removed) State Change CS_REPORTING -> CS_DESTROY 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006(a)public.ip.removed [BREAK] 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1290 Session 5 (sofia/internal/1006(a)public.ip.removed) Locked, Waiting on external entities 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1308 Session 5 (sofia/internal/1006(a)public.ip.removed) Ended 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1310 Close Channel sofia/internal/1006(a)public.ip.removed [CS_DESTROY] 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/1006(a)public.ip.removed) Callstate Change HANGUP -> DOWN 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/1006(a)public.ip.removed) Running State Change CS_DESTROY 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1006(a)public.ip.removed) State DESTROY 2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363 sofia/internal/1006(a)public.ip.removed SOFIA DESTROY 2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570 unmapped public port 31484 protocol UDP to localport 31484 2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570 unmapped public port 31485 protocol UDP to localport 31485 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1006(a)public.ip.removed Standard DESTROY 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1006(a)public.ip.removed) State DESTROY going to sleep The above example was from an externally situated SIP phone ringing voicemail (4000) on FreeSWITCH. I have experimented changing various timers and timeouts in the config of FreeSWITCH (one at a time, being careful to put them back afterwards!) but been unable to resolve the issue. Incidentally, we have no long term intention of running off-site SIP phones with the PBX and I'm hoping not to have to leave it in the DMZ either, it's just like that for debugging. What is a real issue is the calls to our external SIP provider (i.e. outbound calls) being dropped. Any suggestions would be greatly appreciated. Thanks, Matthew Ralston Web Developer & IT Consultant matt(a)mralston.co.uk www.mralston.com

6 8

DISA, originate & bridge
by Boris Kovalenko 14 Jul '11

14 Jul '11

Hello! I'm trying to modify DISA javascript example to use 'originate' instead of 'transfer'. So the problem is - after originated call is answered I see no voice. Thinking this is becase call is not bridged to the current session. How may I do this? My peace of code: osession = new Session("{origination_caller_id_number=7343523000 1,v_ats_dstport=50002}sofia/epbx/89979049898XXX@X.X.X.42:5061"); while(osession.ready()) { // Don't know what to write here } -- Regards, Boris

1 0

How to configure for FS playing voice prompt in case the called extension is in a call.
by fieldpeak 14 Jul '11

14 Jul '11

Hi Gurus, Could anyone advise how to realize have FS to play a voice prompt e.g. 'the extension you dialed is busy now, please dial the other exsention' to replacing busy tone when the called extension is in a call... the details is below, When calling to FS, the FS will play IVR "welcome to call us, please input the extension number, for operator please press 0", then the caller input the extension number, in case the extension is in a call, the caller will hear busy tone... it needs the system play "the extension you dailed is busy, please dial the other extension", currently my dial plan as following, could anyone advise how to change based on below dial plan or any other new dail plan can realize it... Thanks a lot! <extension name="Local_Extension"> <condition field="destination_number" expression="^(1\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/>  <action application="set" data="continue_on_fail=false"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <action application="bridge" data="sofia/internal/${dialed_extension}%${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="IVR" > <condition field="destination_number" expression="^888$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="IVR"/> </condition> </extension>  <menu name="IVR" greet-long="C:/VSWITCH/recordings/greeting_tts.wav" greet-short="C:/VSWITCH/recordings/greeting_tts.wav" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" confirm-macro="" confirm-key="" tts-engine="flite" tts-voice="rms" confirm-attempts="3" timeout="10000" inter-digit-timeout="2000" max-failures="3" max-timeouts="3" digit-len="5"> <entry action="menu-exec-app" digits="/(^\*\d{3,5}$|^\d{3,5}$)/" param="transfer $1 XML default"/> </menu> Regards, Charles

3 6

SPA-2102 ring issue
by Joseph L. Casale 13 Jul '11

13 Jul '11

I have an SPA-2102 a client uses that I can't replace with something good. It appears to ring for several seconds after another user answers the incoming call from some other sip phone. The SPA-3102 appears to have some params that look applicable to tune this but those don't appear in this unit. I don't have an easy way to setup a syslog server remotely, so short of driving out to this place, does anyone know off hand what may cause this so I may give it a try? thanks, jlc

1 0