- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch Server getting down
by ovvenkat 03 Jun '11

03 Jun '11

Hi to all, I dont know whether this problem related to PRI line (OR) wanpipe driver ( OR ) PRI Card I use Linux fedora 13 and Sangoma A101 PRI card to handle incoming calls from E1 lines. Everything works fine for some time. After that, I could see cli debug output saying, all the channels are disconnected and then immediately it says that "signaling status changed to UP". After some time, freeswitch getting shutdown saying ftmod_wanpipe.c:967 [s1c31][1:16] Failed to read from sangoma device: No buffer space available (-65) I pastebin debug logs for the same. http://pastebin.freeswitch.org/16432 I am failed to identify the problem. Can you anyone guide me what is going wrong? Regards Venkatesan OV.

2 1

Transferring to a cell phone
by Yungwei Chen 03 Jun '11

03 Jun '11

Hi, I installed freeswitch from a recent snapshot on CentOS 5. I configured a gateway for my SIP provider in /usr/loca/freeswitch/conf/sip_profiles/external/my.xml, and I can make an outbound call to my cell phone using the following javascript snippet. var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234"); Now I am trying to make a call from a land-line to a javascript application in freeswich, which plays a recording and then transfers the call to a cell phone. But transferring the call to a cell phone doesn't work for some reason. Here's how I transfer the call to a cell phone in javascript: session.execute("transfer", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234"); According to freeswitch.log, the problem seems to be that freeswitch cannot find a dialplan that matches the destiantion number, {ignore_early_media=true}sofia/gateway/sip_provider/1231231234. What am I missing here? Thanks.

5 13

Any way to limit the ability to create conference rooms on demand?
by André Rosowski 03 Jun '11

03 Jun '11

Hi there, I'm using Bigbluebutton (online conferencing) with Freeswitch so that users can call a number and then have to enter the "voicebridge" number to be put into a voice conference. The problem, however, is that if a user dialed the wrong "voicebridge" number he will be put into a new conference thus creating one on demand as stated in the describtion of "mod_conference". Is there any way to manually create conference rooms and make them static...not allowing for new conference "rooms" to be created on demand. default.xml in dialplan: <extension name="phone conference"> <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$"> <action application="conference" data="${pin}@wideband"/> </condition> </extension> Thanks for your help : ) Regards, Realdoe -- Omigos Labs UG (haftungsbeschränkt) - Meisenstraße 96 - 33607 Bielefeld Fon: +49 521 2997 200 - Fax: +49 521 2997 101 info(a)omigos.de - www.omigos.de Geschäftsführer: BSc André Rosowski, Benjamin Bittner, Julian Klima Registergericht: Amtsgericht Bielefeld - Registernummer: HRB 40255 USt-IdNr.: 305/5860/1585

3 3

Conference choppy voice
by Nasir Iqbal 03 Jun '11

03 Jun '11

Hi, I have configured a conference on new installation of freeswitch, every thing works fine voice quality is good when lady says that "You are the only one in this conference" and music but as soon as other party get connected and started to talk voice get choppy/noisy nothing can be understood, like if someone says "HELLO" other party listen jerking noisy voice like "---$H^----#E@----%LL^----*O&" I am new to Freeswitch please guide me that how I can fix / troubleshoot this issue or let me know if need some debugging etc... currently my setup is following CentOs-5 32bit Intel(R) Pentium(R) D CPU 2.66GHz [root@localhost conf]# freeswitch -version FreeSWITCH version: 1.0.head (git-66d16d1 2011-05-20 12-49-16 -0500) [root@localhost conf]# uname -a Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan 21 10:44:23 EST 2009 i686 i686 i386 GNU/Linux [root@localhost conf]# free total used free shared buffers cached Mem: 1001664 959668 41996 0 196484 540676 -/+ buffers/cache: 222508 779156 Swap: 2031608 60 2031548 <?xml version="1.0"?> <include> <configuration name="conference.conf"> <caller-controls> <group name="default-keymap"> <control action="vol talk dn" digits="1"/> <control action="vol talk zero" digits="2"/> <control action="vol talk up" digits="3"/> <control action="vol listen dn" digits="4"/> <control action="vol listen zero" digits="5"/> <control action="vol listen up" digits="6"/> <control action="energy dn" digits="7"/> <control action="energy equ" digits="8"/> <control action="energy up" digits="9"/> <control action="deaf mute" digits="*"/> <control action="mute" digits="0"/> </group> </caller-controls> <profiles> <profile name="conference_1"> <param name="rate" value="8000"/>  <param name="interval" value="20"/> <param name="energy-level" value="20"/> <param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/> <param name="caller-controls" value="default-keymap"/> <param name="tts-engine" value="flite"/> <param name="tts-voice" value="kal"/> <param name="muted-sound" value="conference/conf-muted.wav"/> <param name="mute-detect-sound" value=""/> <param name="max-members" value=""/> <param name="max-members-sound" value=""/> <param name="comfort-noise" value="1"/> <param name="suppress-events" value=""/> <param name="verbose-events" value=""/> <param name="unmuted-sound" value="conference/conf-unmuted.wav"/> <param name="alone-sound" value="conference/conf-alone.wav"/> <param name="moh-sound" value="local_stream://moh"/> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/> <param name="kicked-sound" value="conference/conf-kicked.wav"/> <param name="locked-sound" value="conference/conf-locked.wav"/> <param name="is-locked-sound" value="conference/conf-is-locked.wav"/> <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/> <param name="pin-sound" value="conference/conf-pin.wav"/> <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/> </profile> </profiles> </configuration> </include> Thanks in advance Nasir

3 2

skype reverse engineered
by François Delawarde 03 Jun '11

03 Jun '11

Maybe it could be used to improve mod_skypopen?? http://skype-open-source.blogspot.com/2011/06/skype-protocol-reverse-engine… François.

5 8

ptime mismatch confusion
by Michael Gende 03 Jun '11

03 Jun '11

Hello, If you are a codec/ptime person, great. I've been "round and round" with the vendor on this and and not making much headway. We've an FS switch that has two numbers registered with our provider. That provider is, in turn, getting service from another gateway provider of their own. On our FS, one DID works with no issues and has for more than a year. They other recently started having ptime mismatch issues making it unusable. Tried Scrooge setting to no avail. For the daring (and merciful), here's FS output for a "bad" call and then a "good" one (just a little output from the start): 2011-06-02 14:12:06.756170 [DEBUG] switch_core_state_machine.c:397 (sofia/external/8151231234@xxx.xxx.xxx.xxx:5060<http://8154835414@208.94.159.10:5060/>) Running State Change CS_NEW 2011-06-02 14:12:06.756170 [DEBUG] sofia.c:3210 Channel sofia/external/ 8151231234@xxx.xxx.xxx.xxx:5060 <http://8154835414@208.94.159.10:5060/>entering state [received][100] 2011-06-02 14:12:06.756170 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=Acme_UAS 0 1 IN IP4 xxx.xxx.xxx.xxx s=SIP Media Capabilities c=IN IP4 yyy.yyy.yyy.yyy t=0 0 m=audio 52024 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 *a=maxptime:30 * 2011-06-02 14:12:06.756170 [DEBUG] switch_core_state_machine.c:403 (sofia/external/8151231234@xxx.xxx.xxx.xxx:5060<http://8154835414@208.94.159.10:5060/>) State NEW 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3081 *Audio Codec Compare [PCMU:0:8000:30]/[PCMU:0:8000:**20]* 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3092 Bah HUMBUG! Sticking with PCMU@8000h@20i 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:2039 Set Codec sofia/external/8151231234@xxx.xxx.xxx.xxx:5060<http://8154835414@208.94.159.10:5060/>PCMU/8000 20 ms 160 samples 2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf payload to 101 Note that whilst it says its sticking with ptime of 20 (bolded above),* our vendor in-the-middle sees us as still using ptime=30*. Weird. But, on the same switch with different DID, we get the following "good" result: 2011-06-02 14:15:15.135193 [DEBUG] switch_core_state_machine.c:397 (sofia/external/8151231234@xxx.xxx.xxx.xxx:5060<http://8159535620@208.94.157.10:5060/>) Running State Change CS_NEW 2011-06-02 14:15:15.135193 [DEBUG] sofia.c:3210 Channel sofia/external/8151231234@ xxx.xxx.xxx.xxx:5060 <http://8159535620@208.94.157.10:5060/> entering state [received][100] 2011-06-02 14:15:15.135193 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=Acme_UAS 0 1 IN IP4 xxx.xxx.xxx.xxx s=SIP Media Capabilities c=IN IP4 yyy.yyy.yyy.yyy t=0 0 m=audio 52052 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 *a=maxptime:20* 2011-06-02 14:15:15.135193 [DEBUG] switch_core_state_machine.c:403 (sofia/external/8151231234@xxx.xxx.xxx.xxx.10:5060<http://8159535620@208.94.157.10:5060/>) State NEW 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3081 *Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:**20]* 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3092 Bah HUMBUG! Sticking with PCMU@8000h@20i 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:2039 Set Codec sofia/external/8151231234@xxx.xxx.xxx.xxx:5060<http://8159535620@208.94.157.10:5060/>PCMU/8000 20 ms 160 samples 2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf payload to 101 Naturally, the "far end" vendor says they think its a "FreeSwitch problem". Huh? For my part, I'd just like to FORCE a ptime of 20 in all cases, no matter the codec, for incoming calls, either DID. Tried, but so far, no go. A packet capture showed "bad" call coming in with ptime max of 30, we respond that 30 is OK but them re-invite with ptime set to 20, but it doesn't seem to "catch". On the good DID, they proffer ptime max of 30, we respond with ptime of 20 and life is good. Sorry to post again on this. I have put in some hours but still same results so far. Any suggestions much appreciated. Regards, Mike

3 4

Re: [Freeswitch-users] core.db location
by Thomas Mueller 03 Jun '11

03 Jun '11

> Just tried it. Symlink is created but the issue is when freeswitch > starts, it removes the symlink. So how to handle this ? cannot confirm this: * stopped freeswitch * moved core.db to core.db.symlinked * ln -s core.db.symlinked core.db * start freeswitch * file-list: root root 17 Jun 3 12:11 core.db -> core.db.symlinked freeswitch daemon 144384 Jun 3 12:11 core.db.symlinked core.db still symlinked to core.db.symlinked - Thomas

2 1

core.db location
by sharad 03 Jun '11

03 Jun '11

Dear All Is there any way to host only core.db in some other predefined directory ? Let all other db remain in the folder db. Plz advise. Regards Ankur ----- Original Message ----- From: <freeswitch-users-request(a)lists.freeswitch.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Friday, June 03, 2011 2:35 PM Subject: FreeSWITCH-users Digest, Vol 60, Issue 21 > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)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(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)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: domain (Boris Kovalenko) > 2. Re: domain (Sam) > 3. Re: manipulate CID (Sam) > -------------------------------------------------------------------------------- > _______________________________________________ > 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 >

2 1

manipulate CID
by Sam 03 Jun '11

03 Jun '11

Hello, How do i manipulate callerid number ... i have the below statement <action application="set" data="effective_caller_id_number=66${outbound_caller_id_number}"/> here i want to prefix caller id with 66 so the effective caller id becomes 66 + original callerid . using above statement the caller id passed was just 66 to the far end. below is the dialplan i used. <extension name="7001"> <condition field="destination_number" expression="^(7001)" continue="false"> <action application="set" data="call_timeout=20"/> <action application="info" /> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/> <action application="set" data="effective_caller_id_number=66${outbound_caller_id_number}"/> <action application="ring_ready" /> <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx"/> </extension> ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ This is from header i get . From: "" <sip:66@203.153.53.138> ------------------------------------------------------------------------------------------------------------------- Dialplan: FreeTDM/1:29/7001 Action info() Dialplan: FreeTDM/1:29/7001 Action set(continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED) Dialplan: FreeTDM/1:29/7001 Action set(effective_caller_id_number=66${outbound_caller_id_number}) Dialplan: FreeTDM/1:29/7001 Action ring_ready() Dialplan: FreeTDM/1:29/7001 Action bridge({sip_cid_type=rpid}sofia/gateway/outbound/1646xxxxxxx) 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:119 (FreeTDM/1:29/7001) State Change CS_ROUTING -> CS_EXECUTE 2011-06-02 06:43:48.111347 [DEBUG] switch_core_session.c:1116 Send signal FreeTDM/1:29/7001 [BREAK] 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:359 (FreeTDM/1:29/7001) State ROUTING going to sleep 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:320 (FreeTDM/1:29/7001) Running State Change CS_EXECUTE 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:366 (FreeTDM/1:29/7001) State EXECUTE 2011-06-02 06:43:48.111347 [DEBUG] mod_freetdm.c:451 FreeTDM/1:29/7001 CHANNEL EXECUTE 2011-06-02 06:43:48.111347 [DEBUG] switch_core_state_machine.c:157 FreeTDM/1:29/7001 Standard EXECUTE EXECUTE FreeTDM/1:29/7001 set(call_timeout=20) 2011-06-02 06:43:48.111347 [DEBUG] mod_dptools.c:1059 FreeTDM/1:29/7001 SET [call_timeout]=[20] EXECUTE FreeTDM/1:29/7001 info() 2011-06-02 06:43:48.111347 [INFO] mod_dptools.c:1202 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [RINGING] Channel-State-Number: [4] Channel-Name: [FreeTDM/1:29/7001] Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-Call-UUID: [e5b47553-3b9c-4692-beaa-513a3a3356c2] Answer-State: [ringing] Channel-Read-Codec-Name: [PCMA] Channel-Read-Codec-Rate: [8000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [PCMA] Channel-Write-Codec-Rate: [8000] Channel-Write-Codec-Bit-Rate: [64000] Caller-Direction: [inbound] Caller-Username: [FreeTDM] Caller-Dialplan: [XML] Caller-Caller-ID-Number: [8097328707] Caller-ANI: [8097328707] Caller-Destination-Number: [7001] Caller-Unique-ID: [e5b47553-3b9c-4692-beaa-513a3a3356c2] Caller-Source: [mod_freetdm] Caller-RDNIS: [7001] Caller-Channel-Name: [FreeTDM/1:29/7001] Caller-Profile-Index: [2] Caller-Profile-Created-Time: [1306997028110372] Caller-Channel-Created-Time: [1306997028103773] Caller-Channel-Answered-Time: [0] Caller-Channel-Progress-Time: [0] Caller-Channel-Progress-Media-Time: [0] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] Caller-Screen-Bit: [true] Caller-Privacy-Hide-Name: [false] Caller-Privacy-Hide-Number: [false] variable_direction: [inbound] variable_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2] variable_read_codec: [PCMA] variable_read_rate: [8000] variable_write_codec: [PCMA] variable_write_rate: [8000] variable_channel_name: [FreeTDM/1:29/7001] variable_freetdm_span_name: [wp1] variable_freetdm_span_number: [1] variable_freetdm_chan_number: [29] variable_freetdm_bearer_capability: [0] variable_freetdm_bearer_layer1: [3] variable_freetdm_screening_ind: [network-provided] variable_freetdm_presentation_ind: [presentation-allowed] variable_dialed_extension: [7001] variable_transfer_ringback: [local_stream://moh] variable_#: [true] variable_export_vars: [#,*,#] variable_bind_meta_key: [#] variable_hangup_after_bridge: [true] variable_continue_on_fail: [true] variable_called_party_callgroup: [1] variable_dialed_user: [7001] variable_dialed_domain: [192.168.2.190] variable_call_uuid: [e5b47553-3b9c-4692-beaa-513a3a3356c2] variable_originate_disposition: [USER_NOT_REGISTERED] variable_max_forwards: [70] variable_call_timeout: [20] variable_current_application: [info] Regards Sam

3 8

mrcp
by Jan Berger 03 Jun '11

03 Jun '11

Hi, I would like to communicate with MRCP to control TTS and ASR from a C module - how can I do that easiest? Jan

3 5