November 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP Privacy on outbound calls (FS 1.2.13)
by Martyn Davies 13 Nov '13

13 Nov '13

Hello all, I'm experimenting with SIP privacy and trying to emulate the message format used by an operator I'm working with. My originate command (from FS_CLI) looks like this: originate {origination_caller_id_name='anonymous(a)anonymous.invalid ',origination_caller_id_number=631,sip_cid_type=pid,origination_privacy=screen+hide_number+hide_name}sofia/external/ sip:447540xxxxxx@sip.xxxx.com & conference('martyn') Which very nearly works. The receiving mobile says "Unknown" for the caller-id, which is the desired result, but at the SIP level there are a couple of differences from the operator's own format that I wanted to get exactly right. The 'From:' field contains: "anonymous" <sip:631@10.1.1.1> My P-Asserted-Identity field is also the same. A couple of things: (1) Is is possible for the name to get reported as "anonymous(a)anonymous.invalid" as the caller name, just as I specified, instead of truncated at the @ ? (2) I'm happy with the P-Asserted-Identity field containing the caller number, but can this be removed from the "From" field? When I receive calls from the operator, there is the anonymous name, but the caller number is missing. Regards, Martyn

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luamod won't compile under lua 5.2
by Muhammad Naseer Bhatti 13 Nov '13

13 Nov '13

Hi, Someone got a chance to look into FS-5649? -- Thanks, Muhammad Naseer Bhatti

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problem with group call
by Jason Holden 13 Nov '13

13 Nov '13

I currently have a freeswitch running 1.3.5 and am having the following problem. When bridging to a group only the first extension in the directory is rining. I've tried added the +A flag with no success. Any suggestions on how I can have all pointers in the group call ring?

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Re: [Freeswitch-users] Audio Quality of recordings
by Scott Ross 13 Nov '13

13 Nov '13

I wasn't advised to use ubuntu, just including it as another data point (its my dev machine). I've tested against two physical servers. RedHat 6, Xeon 6 core 2.6ghz, 32g ram The dev machine running Ubuntu 13.04, decent Core i7, 12g ram I've tested on several XenServer instances running RedHat 6. All devises/hosts are on the same internal network. Seeing the same results regardless. The device on the other end of the calls is an external conference bridge (physical host). There could be some device specific issues - but I see the same results when testing against another freeswitch instance on the same network set up with mod_conference. Using g711 and recording to .wav. The call sounds unsatisfactory in real time and in the recordings Call quality is near perfect when the two calls aren't bridged into a freeswtich conference or our external conference bridge. But even setting up a basic freeswitch conference to run the tests against resulted in the lower quality. So "failure" case appears to be when two separately originated calls being bridged into any single conferenced environment. Just to be clear - the quality is ok, but not great. For reference, I'm expecting a PESQ score of 3.8 or greater - which is met with asterisk (seeing 3.8 - 4.3), but Freeswitch in its current setup consistently results in 0.5 - 1 point lower at around 2.9 - 3.7. I'm hoping there is some environment/configuration tweaking I can do to fix this. I have tried running FS with the -hp flag. Do I need to make configuration changes on my OS/network to account for that? Does this feel like an OS level issue? Do I need to try Debian 6? Thank you for the reply. Again, let me know what other info I can provide.

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reroute conference audio
by Joli Martinez 13 Nov '13

13 Nov '13

Hello, I would like to send the conference bridge audio to another sip server. So basically Caller calls freeswitch free switch places user to conference1 and then sends audio stream via SIP to a separate SIP server which will then record and keep track of all the reporting for the conference. I know freeswitch can do the recording and reporting, but there is a dedicated box already for that and that is what y boss wants. Could someone give me some suggestion as to how I can accomplish this? Thanks,

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TEST
by Brian West 12 Nov '13

12 Nov '13

-- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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spaces in params to uuid_broadcast
by Alex Lake 12 Nov '13

12 Nov '13

I want to use uuid_broadcast to run a lua app with a few parameters. http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast Doesn't give an example of this, and the few things I've tried didn't really work. This isn't even worth bothering with: uuid_broadcast f6db28c5-1703-4819-a285-d4b197983bb6 lua::mcm3.lua param1 param2 bleg This one didn't seem to work: uuid_broadcast f6db28c5-1703-4819-a285-d4b197983bb6 "lua::mcm3.lua param1 param2" bleg Any clues? ;-)

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Nibblebiil "Pause" causing session locked
by doremo chin 12 Nov '13

12 Nov '13

Hi All, I'm trying to use nibblebill "pause" function for a particular call but the call session locked after the call hangup. Dialplan, <extension name="outofcredit"> <condition field="destination_number" expression="^(outofcredit)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="nibble_rate=0"/> <action application="nibblebill" data="pause"/> <action application="answer"/> <action application="playback" data="$${sounds_dir}/en/us/callie/misc/8000/misc-your_call_has_been_terminated.wav"/> <action application="hangup"/> </condition> </extension> Call logs, 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:48 sofia/sipinterface_2/4668(a)192.168.0.106 Standard HANGUP, cause: NONE 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:676 (sofia/sipinterface_2/4668(a)192.168.0.106) State HANGUP going to sleep 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:689 (sofia/sipinterface_2/4668(a)192.168.0.106) Callstate Change ACTIVE -> HANGUP 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:446 (sofia/sipinterface_2/4668(a)192.168.0.106) State Change CS_HANGUP -> CS_REPORTING 2013-07-30 20:38:33.360312 [DEBUG] switch_core_session.c:1340 Send signal sofia/sipinterface_2/4668(a)192.168.0.106 [BREAK] 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_2/4668(a)192.168.0.106) Running State Change CS_REPORTING 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:761 (sofia/sipinterface_2/4668(a)192.168.0.106) State REPORTING 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:92 sofia/sipinterface_2/4668(a)192.168.0.106 Standard REPORTING, cause: NONE 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:761 (sofia/sipinterface_2/4668(a)192.168.0.106) State REPORTING going to sleep 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:440 (sofia/sipinterface_2/4668(a)192.168.0.106) State Change CS_REPORTING -> CS_DESTROY 2013-07-30 20:38:33.360312 [DEBUG] switch_core_session.c:1340 Send signal sofia/sipinterface_2/4668(a)192.168.0.106 [BREAK] 2013-07-30 20:38:33.360312 [DEBUG] switch_core_session.c:1548 Session 3 (sofia/sipinterface_2/4668(a)192.168.0.106) Locked, Waiting on external entities show channels & uuid_kill, freeswitch@internal> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num f161cb1c-f914-11e2-92d1-35a97464e755,inbound,2013-07-30 20:38:31,1375187911,sofia/sipinterface_2/4668@192.168.0.106 ,CS_EXECUTE,4668,4668,192.168.0.133,0,playback,/usr/local/freeswitch/sounds/en/us/callie/misc/8000/misc-your_call_has_been_terminated.wav,XML,out_bound,PCMA,8000,64000,PCMA,8000,64000,, sip.wit.net.my,4668(a)192.168.0.106,,ACTIVE,,,,,, 1 total. freeswitch@internal> uuid_kill f161cb1c-f914-11e2-92d1-35a97464e755 -ERR No such channel! freeswitch@internal> May i know is there any way to release the locked session? or is it a bug in freeswitch ? Thanks, dc

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mute/unmute conference
by Joli Martinez 12 Nov '13

12 Nov '13

Hello, I need the ability to have first person to join the conference have the ability to mute/unmute the other callers. Any idea how to acomplish this. Thanks,

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OPUS parameters
by Jan Sverre Andreassen 11 Nov '13

11 Nov '13

I want to control OPUS bandwidth and sound quality. Is it an easy way to send parameters to mod_opus? Best Regards, Jan Sverre Andreassen ________________________________ This email and any files contained therein is confidential and may contain privileged information. If you are not the named addressee(s) or you have otherwise received this in error, you should not distribute or copy this e-mail or use any of its content for any purpose. Please notify the sender immediately by e-mail if you have received this e-mail in error and delete it from your system.

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