March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_dingaling and google voice
by envelopes envelopes 02 Mar '11

02 Mar '11

Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call. is there any config variable to set up so that I will be notified if my call is answered? thanks!

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Remaking default config files in Centos
by Eric Phipps 02 Mar '11

02 Mar '11

Hello, I'm working with Freeswitch and have managed to place an extra "<" somewhere in my xml files. Unfortunately, I have been unable to locate it and it's preventing the starting of Freeswitch. I was curious if there was an easy way to overwrite the configs which exist there now and replace them with the default configurations for CentOS. -Eric Phipps

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FreeTDM errors in log file
by Stephen Wilde 02 Mar '11

02 Mar '11

I'm using Sangoma A108 board in my FS installation and my log is full of warnings as: 2011-03-01 00:18:14.947380 [WARNING] mod_freetdm.c:434 [s17c22][17:22] VETO state change from PROGRESS_MEDIA to PROCEED 2011-03-01 00:18:53.151673 [WARNING] mod_freetdm.c:434 [s3c11][3:11] VETO state change from PROGRESS_MEDIA to PROCEED 2011-03-01 00:19:23.001140 [WARNING] mod_freetdm.c:434 [s23c29][23:29] VETO state change from RINGING to PROCEED 2011-03-01 00:19:40.862103 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO state change from PROGRESS_MEDIA to PROCEED 2011-03-01 00:19:46.022154 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO state change from PROGRESS_MEDIA to PROCEED 2011-03-01 00:19:50.608224 [WARNING] mod_freetdm.c:434 [s22c24][22:24] VETO state change from PROGRESS_MEDIA to PROCEED 2011-03-01 00:19:52.422221 [WARNING] mod_freetdm.c:434 [s3c21][3:21] VETO state change from PROGRESS_MEDIA to PROCEED 2011-03-01 00:21:36.461338 [WARNING] mod_freetdm.c:434 [s3c31][3:31] VETO state change from PROGRESS_MEDIA to PROCEED 2011-03-01 00:21:36.981455 [WARNING] mod_freetdm.c:434 [s3c30][3:30] VETO state change from PROGRESS_MEDIA to PROCEED Another kind of messages that are in my log file are: 2011-03-01 00:53:14.427853 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760 [SNGISDN Q931] s16: Protocol: Unknown Event Code(2): Incomp Msg(276) 2011-03-01 00:53:18.428846 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557 [s16c23][16:23] Received DISCONNECT in an invalid state (TERMINATING) 2011-03-01 00:53:18.428846 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562 [s16c23][16:23] Why bother changing state from TERMINATING to TERMINATING 2011-03-01 00:54:08.607351 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760 [SNGISDN Q931] s17: Protocol: Unknown Event Code(2): Incomp Msg(276) 2011-03-01 00:54:12.607389 [CRIT] ftmod_sangoa_isdn_stack_hndl.c:557 [s17c20][17:20] Received DISCONNECT in an invalid state (TERMINATING) 2011-03-01 00:54:12.607389 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562 [s17c20][17:20] Why bother changing state from TERMINATING to TERMINATING 2011-03-01 00:56:45.595869 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760 [SNGISDN Q931] s16: Protocol: Unknown Event Code(2): Incomp Msg(276) 2011-03-01 00:56:49.596913 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557 [s16c23][16:23] Received DISCONNECT in an invalid state (TERMINATING) 2011-03-01 00:56:49.596913 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562 [s16c23][16:23] Why bother changing state from TERMINATING to TERMINATING 2011-03-01 00:59:23.036314 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:760 [SNGISDN Q931] s17: Protocol: Unknown Event Code(2): Incomp Msg(276) 2011-03-01 00:59:27.037327 [CRIT] ftmod_sangoma_isdn_stack_hndl.c:557 [s17c20][17:20] Received DISCONNECT in an invalid state (TERMINATING) 2011-03-01 00:59:27.037327 [WARNING] ftmod_sangoma_isdn_stack_hndl.c:562 [s17c20][17:20] Why bother changing state from TERMINATING to TERMINATING I'm using latest git of Freeswitch, Wanpipe 3.5.18 and lib Sangoma isdn 7.3.0. Attached to this email the .pcap related to the call that generate the critical error (to me it seems that is missed a CALL_PROCEEDING message). Stephen

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FreeSWITCH Conference Call Today
by Michael Collins 02 Mar '11

02 Mar '11

Hello all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_02 We are going to have an update from Mark Crane on the FusionPBX project. Bring your questions! -MC

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LCR troubles
by Serge S. Yuriev 02 Mar '11

02 Mar '11

Hello, I have created JIRA about this FS-3109. I post it here to wide it's coverage. In short: in LCR mode FS forwards (negative) GW answer to caller but continues on next GW so useless one legged call created. If I'm doing this via list of gateways - callflow continues normal. Pls advice. -- wbr, Serge

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Eavesdrop - How to get out of the application ?
by keithmsn＠laaks.com 02 Mar '11

02 Mar '11

Hi, I am calling the eavesdrop application as follows: session.setVariable("eavesdrop_enable_dtmf","true"); session.execute("eavesdrop","all"); Works 100%, with the application responding to the keys for talking to the one, other or both sides of the channel, resetting and also moving onto the next channel (*). But there is just no way to get out of eavesdrop other than hanging up the call and calling back in. Maybe I am missing something? I looked into: mod_dptools.c at 'eavesdrop_function', switch_ivr_async.c at 'switch_ivr_eavesdrop_session' but no code for some escape key (i.e. #) in 'switch_ivr_eavesdrop_session' which will break the permanent loop in 'eavesdrop_function'. Maybe I should not be using 'all' and instead go the db route? I also spotted two variables: "eavesdrop_announce_id" and "eavesdrop_annnounce_macro" used in switch_ivr_eavesdrop_session. Looks like this allows for a macro to be played on switching to a new session to verbally identify it. Don't see this in the wiki. Anybody tried this before? Regards Keith

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Does someone has a Huawei K3765 runing with FreeSWITCH?
by Michelle Konzack 02 Mar '11

02 Mar '11

Hello, I have a Vodafone DSL EasyBox 803A (Astoria Networks) with the Huawei K3765 runing and it works, but since the EasyBoy 803A does not support SIP, I want to drop it and use the Huawei K3765 directly with FreeSWITCH Does someone has the Huawei K3765 runing with FreeSWITCH? And how canI configure it without lossing of the Internet connection? Thanks, Greetings and nice Day/Evening Michelle Konzack -- ##################### Debian GNU/Linux Consultant ###################### Development of Intranet and Embedded Systems with Debian GNU/Linux itsystems@tdnet France EURL itsystems@tdnet UG (limited liability) Owner Michelle Konzack Owner Michelle Konzack Apt. 917 (homeoffice) 50, rue de Soultz Kinzigstraße 17 67100 Strasbourg/France 77694 Kehl/Germany Tel: +33-6-61925193 mobil Tel: +49-177-9351947 mobil Tel: +33-9-52705884 fix <http://www.itsystems.tamay-dogan.net/> <http://www.flexray4linux.org/> <http://www.debian.tamay-dogan.net/> <http://www.can4linux.org/> Jabber linux4michelle(a)jabber.ccc.de Linux-User #280138 with the Linux Counter, http://counter.li.org/

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Changing voicemail greeting to pronounce (instead of spelling out) id/number
by Johan Helsingius 02 Mar '11

02 Mar '11

Hi! I am using actual names of users instead of numbers (numbers are so POTS), but there doesn't seem to be a way to change the way the voicemail user id is announced - I know you can use vm-alternate-greet-id to change the ID, but if it is not numerical, it gets spelled out letter by letter instead of pronounced. Is there any way to change this behavior without changing the source code? Julf

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How to use uuid_chat and uuid_display?
by mayamatakeshi 02 Mar '11

02 Mar '11

What are the conditions for uuid_chat and uuid_display to work? I have two channels bridged (using intercept) but when i try with them in the FS cli, although I get a positive reply, nothing is sent to the SIP agents (i have enabled siptrace but also confirmed using tcpdump): freeswitch@IPX050> sofia loglevel nua 9 Sofia log level for component [nua] has been set to [9] freeswitch@IPX050> sofia profile internal siptrace on nua: nua_set_params: entering nua((nil)): sent signal r_set_params Enabled sip debugging on internal nua((nil)): recv signal r_set_params nua: nua_stack_set_params: entering nua((nil)): event r_set_params 200 OK nua: nua_application_event: entering freeswitch@IPX050> uuid_chat 19d300fa-3d95-4b09-a2a7-2ffefa708b3c test +OK 2011-03-02 16:59:23.665054 [DEBUG] switch_core_session.c:876 Send signal sofia/internal/term6(a)dev2.basix5.ne.jp [BREAK] freeswitch@IPX050> uuid_display 19d300fa-3d95-4b09-a2a7-2ffefa708b3c test +OK Success The header Allow in the INVITE from the terminals is like this: Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE I have also tried application send_display in the dialplan but nothing happens. br, takeshi

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HANGUP_CAUSE after bridge in outbound ESL
by Dmitry Sytchev 02 Mar '11

02 Mar '11

Hi All What is correct method to get bridge status after execute("bridge") in ESL? I have inbound call that gets bridged to SIP endpoint. I need to know whether it was ORIGINATOR_CANCEL or BUSY or something else, but if I use sync, seems I can't determine ORIGINATOR_CANCEL status, because it is set on a-leg which is destroyed first after hangup. Is there a way to get bridge status without messing with events and event source filtering by channel uuid or event type? -- Best regards, Dmitry Sytchev, IT Engineer

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