March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] last svn version compile error
by Madovsky 02 Mar '10

02 Mar '10

The compilation is fine if I comment out mod_vmd from modules.conf Thanks F ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 02, 2010 1:24 AM Subject: last svn version compile error Hi, Ino succes to update freeswitch to version 16859 with usual command "make current" etc... but now I got this error : /usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: passing argument 6 of âswitch_core_media_bug_addâ makes integer from pointer without a cast /usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: too few arguments to function âswitch_core_media_bug_addâ Any idea ? thanks Franck Linux Fedora 10 64bits

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How to tell if a call timed out
by Mark Sobkow 02 Mar '10

02 Mar '10

I'm using the originate command from Erlang, specifying a timeout of 30 seconds. It is properly giving up on dialing, but it's still returning a call UUID instead of an error/timed out status. How can I query Freeswitch to determine whether the call was answered or timed out, without resorting to the event interface? (I could put together a dialplan that calls the event interface after doing the originate, but from what I've seen that event interface won't execute if the previous command "fails", such as when I did a play_and_get_digits timing out.) -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com

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Inbound settings
by Scott Fernandez 02 Mar '10

02 Mar '10

Hi, 1) How to configure inbound settings in FS? 2) If so, Please list down the steps that I require to do? 3) If have more than 1 DID number (let;'s say 4 DID's

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Freeswitch/Openzap dials out on disconnected FXO line
by Robert Hadley 02 Mar '10

02 Mar '10

When dialing out, Freeswitch/Openzap is not detecting that an analog FXO channel is disconnected and tries dialing out on the channel anyway. No error is reported. The call doesn't timeout until a minute later. Shouldn't Freeswitch/Openzap skip over a disconnected channel to the next connected channel? I have configured a Sangoma A200 FXO card as a FXO span. [span wanpipe FXO] name => PSTN Line 1 number => 4253491059 fxo-channel => 2:3 name => PSTN Line 2 number => 4253491058 fxo-channel => 2:4 The wanpipe driver does detect and report when a CO line is connected or disconnected (in /var/log/messages), and Freeswitch/Openzap gets an event as reported in the log. /var/log/messages: Feb 25 15:23:10 roberth-c53 kernel: wanpipe2: Module 3: FXO Line is disconnected! FS_CLI: 2010-02-25 15:23:10.711604 [DEBUG] ozmod_analog.c:788 EVENT [ALARM_TRAP][3:1] STATE [DOWN] /var/log/messages: Feb 25 15:23:44 roberth-c53 kernel: wanpipe2: Module 4: FXO Line is connected! FS_CLI: 2010-02-25 15:23:44.901979 [DEBUG] ozmod_analog.c:788 EVENT [ALARM_CLEAR][3:2] STATE [DOWN] I have the dialplan configured to use the next available port in the FXO span (there will be more than 2 channels later). <extension name="OutgoingFXO"> <condition field="destination_number" expression="^7(\d+)$"> <action application="bridge" data="openzap/FXO/a/$1"/> </condition> </extension> Here is a portion of the log when that shows dialing out on a disconnected analog FXO channel. EXECUTE sofia/internal/5410@192.168.72.45:5060 bridge(openzap/FXO/a/93491045) 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1257 Connect outbound channel OpenZAP/3:1/93491045 2010-02-25 15:26:17.891443 [NOTICE] switch_channel.c:642 New Channel OpenZAP/3:1/93491045 [3c8f46f5-77a8-498f-a51c-015837746cb7] 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:1269 (OpenZAP/3:1/93491045) State Change CS_NEW -> CS_INIT 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal OpenZAP/3:1/93491045 [BREAK] 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:59 Changing state on 3:1 from DOWN to DIALING 2010-02-25 15:26:17.891443 [WARNING] switch_core_session.c:486 OpenZAP/3:1/93491045 does not support the proxy feature, disabling. 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread starting. 2010-02-25 15:26:17.891443 [DEBUG] ozmod_analog.c:450 Executing state handler on 3:1 for DIALING 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/3:1/93491045) Running State Change CS_INIT 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/3:1/93491045) State INIT 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:394 (OpenZAP/3:1/93491045) State Change CS_INIT -> CS_ROUTING 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal OpenZAP/3:1/93491045 [BREAK] 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/3:1/93491045) State INIT going to sleep 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/3:1/93491045) Running State Change CS_ROUTING 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/3:1/93491045) State ROUTING 2010-02-25 15:26:17.891443 [DEBUG] mod_openzap.c:417 OpenZAP/3:1/93491045 CHANNEL ROUTING 2010-02-25 15:26:17.891443 [DEBUG] switch_ivr_originate.c:66 (OpenZAP/3:1/93491045) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-02-25 15:26:17.891443 [DEBUG] switch_core_session.c:1019 Send signal OpenZAP/3:1/93491045 [BREAK] 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/3:1/93491045) State ROUTING going to sleep 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/3:1/93491045) Running State Change CS_CONSUME_MEDIA 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/3:1/93491045) State CONSUME_MEDIA 2010-02-25 15:26:17.891443 [DEBUG] switch_core_state_machine.c:360 (OpenZAP/3:1/93491045) State CONSUME_MEDIA going to sleep Thanks, Robert

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Re: [Freeswitch-users] SQL ERR: library routine called out of sequence
by Anthony Minessale 02 Mar '10

02 Mar '10

You probably had it right in the last email. Polling the core's db from lua is not recommended. We use sla extensively and never once see your issue. On Feb 28, 2010 8:55 AM, "Yehavi Bourvine" <yehavi.bourvine(a)gmail.com> wrote: After a few more tests I *think* it is related to SLA. Since it cannot be reproduced consistenty I say "think". What I have is: - 80635 which is an SLA extension between two Polycoms. - 80636 which is a private extension on Polycom - 80632 which is a Cisco extension. - 86111 which is connected behind a Cisco SIP<->PSTN gateway. Since this gateway doesn't hav any login information it is not defined as a gateway but accepted via ACL. The tests that passed ok: - From/to 80636/80632 to 80635 (i.e. - all internal). - From/to 80635 to 80636/80632 (i.e. - all internal) - From/to 80636/80632 to 86111 and vice versa (i.e. via the gateway, single extensions). The one that fails after a few attempts: - From 86111 to 80635 (i.e. from the gateway to a shared extension). I hope that this gives some more clues. Thanks, __Yehavi: 2010/2/28 Yehavi Bourvine <yehavi.bourvine(a)gmail.com> > > Hello, > > I've installed a machine with CentOS-5.4 and the latest FreeSwitch (16841M). The... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Freeswitch & T.38
by Sharad 02 Mar '10

02 Mar '10

Hi friends, Is Freeswitch 1.05 carrying T.38 support also ? If not, do we have any community member who can provide paid T.38 support with Freeswitch ? -- View this message in context: http://n2.nabble.com/Freeswitch-T-38-tp4660389p4660389.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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ivr from mysql
by michal kalinowski 02 Mar '10

02 Mar '10

Hi, I need build ivr script/aplication which will take dynamically configuration from mysql db (ivr menu, prompts, etc.). My first idea is generate xml from python script. But it's not working properly. Anybody has some idea or have this aplication already done ? BR, Michał

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last svn version compile error
by Madovsky 02 Mar '10

02 Mar '10

Hi, Ino succes to update freeswitch to version 16859 with usual command "make current" etc... but now I got this error : /usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: passing argument 6 of âswitch_core_media_bug_addâ makes integer from pointer without a cast /usr/src/freeswitch.trunk/src/mod/applications/mod_vmd/mod_vmd.c:714: error: too few arguments to function âswitch_core_media_bug_addâ Any idea ? thanks Franck Linux Fedora 10 64bits

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Re: [Freeswitch-users] Announcement: FreeSWITCH Add Broadsoft SCA Support
by Anthony Minessale 02 Mar '10

02 Mar '10

You have to set the param to sylantro to get that mode. Or configure your phone to use broadsoft. On Feb 17, 2010 7:54 AM, "Yehavi Bourvine" <yehavi.bourvine(a)gmail.com> wrote: I am trying to migrate from 1.0.5pre10 to the latest (FreeSWITCH Version 1.0.5-20100216-0400 (16659M)), and shared apearance stopped working... We have Polycom phones and I use the same configuration for both versions. With the old version I see that after the phone registers with FreeSwitch, the server subscribes to the phone for the extension; with the latest version this does not happen. Furthermore, the table sip_shared_appearance_dialogs is empty. I don't find anything I can change on the phone config (the line is already set to shared). Here is the relevant config from the sip profile: <param name="manage-presence" value="passive"/> (set to TRUE only on one profile). <param name="manage-shared-appearance" value="true"/> <param name="dbname" value="share_presence"/> <param name="presence-hosts" value="pbx-gr-backup.ip-phones.huji.ac.il "/> Any idea where to look? Thanks, __Yehavi: _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Re: [Freeswitch-users] smg_prid not bridging the call
by Davud Yat Sin 02 Mar '10

02 Mar '10

Hi lakshmanan, This was fixed in Openzap revision:1047 Can you update your openzap and update to these wanpipe drivers: ftp://ftp.sangoma.com/linux/custom/DavidYS/wanpipe-3.5.8.6.smg_pri-v1.63.tgz David Yat Sin, BEng, Software Engineer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 119 | e. dyatsin(a)sangoma.com www.sangoma.com | wiki.sangoma.com Dear all, I'm having a A102 Sangoma hardware. I configured it with freeswitch. wanrouter status, says both the port as connected. My smg_prid version is Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: ================System restart============= Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Sangoma PRI Protocol Stack Daemon = Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Version: 1.54 = Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Date: Feb 15 2010 = Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Wanpipe Release: wanpipe-3.5.8.6 = Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: = Revision:Revision: 15288 = Feb 26 16:08:14 FMS-FreeSwitch sangoma_prid: =========================================== My freeswitch version is 16729. I started freeswitch. oz list +OK span: 1 (smg_prid) type: Sangoma (boost) chan_count: 60 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none I originated a call as originate openzap/smg_prid/a/9952248266 &park(), which hits my mobile. But when I issued the following command: originate openzap/smg_prid/a/9952248266 &bridge(openzap/smg_prid/a/8122133885) It rings my mobile (9952248266) first, but after that the following error was displayed 2010-02-26 16:20:51.736080 [ERR] switch_ivr_originate.c:2387 Cannot create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION] The call got ended in my mobile. Freeswitch log and smg_pri.conf http://pastebin.freeswitch.org/12248 openzap.conf: [span wanpipe smg_prid] name => smg_prid trunk_type =>e1 b-channel => 1:1-15 b-channel => 1:17-31 trunk_type =>e1 b-channel => 2:1-15 b-channel => 2:17-31 openzap.conf.xml: <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <boost_spans> <span id="smg_prid"> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </boost_spans> </configuration> Please guide me to setup this one!!.

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