February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Help with early end to ringback when calling PSTN?
by Harry Coin 26 Feb '12

26 Feb '12

Hello all, great project! In the simple case of dialing out through a small 4 port fxo (grandstream-4104) there is an unwelcome ringback silence gap of almost exactly 10 seconds while the fxo goes offhook and issues its dtmf, then the audio resumes with whatever audio the PSTN sends in response to the dialled number. I've tried to continue the ringback music using ignore_early_media=true with no success. Tried transfer_ringback, instant_ringback, no joy. In the following short fs_cli trace no matter what I try the music stops at the point below noted as 12:19:44... and there is silence until 12:19:54... when the PSTN provides audio. What can I do to keep the ringback music going until the log point noted at ...:54xxxx below? 2012-02-23 12:19:44.000317 [DEBUG] mod_dptools.c:1176 sofia/internal/011(a)gate.quietfountain.com SET [bypass_media]=[true] EXECUTE sofia/internal/011(a)gate.quietfountain.com bridge({ignore_early_media=true}sofia/internal/9935551212@fxoports1to4.quietfountain.com|sofia/internal/9915551212@fxoports1to4.quietfountain.com) 2012-02-23 12:19:44.000317 [DEBUG] switch_ivr_originate.c:1882 Parsing global variables 2012-02-23 12:19:44.000317 [DEBUG] switch_event.c:1521 Parsing variable [ignore_early_media]=[true] 2012-02-23 12:19:44.000317 [NOTICE] switch_channel.c:911 New Channel sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [ab71adf5-4a5e-e111-b2c4-00104b98432d] 2012-02-23 12:19:44.000317 [DEBUG] mod_sofia.c:4496 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State Change CS_NEW -> CS_INIT 2012-02-23 12:19:44.000317 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) Running State Change CS_INIT 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State INIT 2012-02-23 12:19:44.021687 [DEBUG] mod_sofia.c:85 sofia/internal/9935551212(a)fxoports1to4.quietfountain.com SOFIA INIT 2012-02-23 12:19:44.021687 [DEBUG] mod_sofia.c:125 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State Change CS_INIT -> CS_ROUTING 2012-02-23 12:19:44.021687 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State INIT going to sleep 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) Running State Change CS_ROUTING 2012-02-23 12:19:44.021687 [DEBUG] switch_channel.c:1840 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) Callstate Change DOWN -> RINGING 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State ROUTING 2012-02-23 12:19:44.021687 [DEBUG] mod_sofia.c:148 sofia/internal/9935551212(a)fxoports1to4.quietfountain.com SOFIA ROUTING 2012-02-23 12:19:44.021687 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-02-23 12:19:44.021687 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State ROUTING going to sleep 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) Running State Change CS_CONSUME_MEDIA 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State CONSUME_MEDIA 2012-02-23 12:19:44.021687 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/9935551212(a)fxoports1to4.quietfountain.com) State CONSUME_MEDIA going to sleep 2012-02-23 12:19:44.021687 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] send 1397 bytes to udp/[192.168.29.55]:5060 at 18:19:44.034612: ------------------------------------------------------------------------ INVITE sip:9935551212@fxoports1to4.quietfountain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.29.1;rport;branch=z9hG4bKQKjKXKZBmD1Sr Max-Forwards: 69 From: "Net Ops Desk L1" <sip:x011@192.168.29.1>;tag=3F61avHXm0t2K To: <sip:9935551212@fxoports1to4.quietfountain.com> Call-ID: cd0d8ada-d8ed-122f-669c-3fb68015248b CSeq: 24674528 INVITE Contact: <sip:mod_sofia@192.168.29.1:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7f17ad7 2011-10-01 14-05-14 +0200 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 445 X-FS-Support: update_display P-Asserted-Identity: "Net Ops Desk L1" <sip:x011@192.168.29.1> v=0 o=011 8004 8000 IN IP4 192.168.29.57 s=SIP Call c=IN IP4 192.168.29.57 t=0 0 m=audio 5012 RTP/AVP 0 8 4 18 112 97 103 102 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:103 AAL2-G726-40/8000 a=rtpmap:102 G729E/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32-36,54 a=ptime:20 ------------------------------------------------------------------------ 2012-02-23 12:19:44.021687 [DEBUG] sofia.c:5174 Channel sofia/internal/9935551212(a)fxoports1to4.quietfountain.com entering state [calling][0] recv 353 bytes from udp/[192.168.29.55]:5060 at 18:19:44.040757: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.29.1;rport;branch=z9hG4bKQKjKXKZBmD1Sr From: "Net Ops Desk L1" <sip:x011@192.168.29.1>;tag=3F61avHXm0t2K To: <sip:9935551212@fxoports1to4.quietfountain.com> Call-ID: cd0d8ada-d8ed-122f-669c-3fb68015248b CSeq: 24674528 INVITE User-Agent: Grandstream GXW4104 (HW 1.0, Ch:2) 1.3.4.10 Content-Length: 0 ------------------------------------------------------------------------ SNOOZE WITH UNWELCOME SILENCE..... recv 503 bytes from udp/[192.168.29.55]:5060 at 18:19:54.350776: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.29.1;rport;branch=z9hG4bKQKjKXKZBmD1Sr From: "Net Ops Desk L1" <sip:x011@192.168.29.1>;tag=3F61avHXm0t2K To: <sip:9935551212@fxoports1to4.quietfountain.com>;tag=7be5ae114be4f046 Call-ID: cd0d8ada-d8ed-122f-669c-3fb68015248b CSeq: 24674528 INVITE User-Agent: Grandstream GXW4104 (HW 1.0, Ch:2) 1.3.4.10 Contact: <sip:192.168.29.55:5060;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 ------------------------------------------------------------------------ 2012-02-23 12:19:54.332358 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:54.332358 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:54.353976 [DEBUG] sofia.c:5174 Channel sofia/internal/9935551212(a)fxoports1to4.quietfountain.com entering state [proceeding][180] 2012-02-23 12:19:54.353976 [NOTICE] sofia.c:5253 Ring-Ready sofia/internal/9935551212(a)fxoports1to4.quietfountain.com! 2012-02-23 12:19:54.353976 [NOTICE] mod_sofia.c:2393 Ring-Ready sofia/internal/011(a)gate.quietfountain.com! recv 797 bytes from udp/[192.168.29.55]:5060 at 18:19:54.362180: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.29.1;rport;branch=z9hG4bKQKjKXKZBmD1Sr From: "Net Ops Desk L1" <sip:x011@192.168.29.1>;tag=3F61avHXm0t2K To: <sip:9935551212@fxoports1to4.quietfountain.com>;tag=7be5ae114be4f046 Call-ID: cd0d8ada-d8ed-122f-669c-3fb68015248b CSeq: 24674528 INVITE User-Agent: Grandstream GXW4104 (HW 1.0, Ch:2) 1.3.4.10 Contact: <sip:192.168.29.55:5060;transport=udp> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Supported: replaces, timer, 100rel, path Content-Length: 224 v=0 o=system 8002 8000 IN IP4 192.168.29.55 s=SIP Call c=IN IP4 192.168.29.55 t=0 0 m=audio 5012 RTP/AVP 0 8 4 18 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 ------------------------------------------------------------------------ 2012-02-23 12:19:54.353976 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:54.353976 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/9935551212(a)fxoports1to4.quietfountain.com [BREAK] 2012-02-23 12:19:54.353976 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/011(a)gate.quietfountain.com [BREAK] 2012-02-23 12:19:54.353976 [NOTICE] switch_ivr_originate.c:481 Ring Ready sofia/internal/011(a)gate.quietfountain.com!

3 3

bypass_media->true state causes sofia bridge under loopback to fail
by Harry Coin 26 Feb '12

26 Feb '12

I'm thinking this is a bug: A local extension calls internal dialplan ring group. If the local extension calls the internal dialplan at 1234 directly, whether bypass_media=true or bypass_media=false it works. However when that same extension is called via bridge loopback/1234 in a ring group it fails with 'normal temporary failure' only when bypass_media=true; it works when bypass_media is false. What am I missing? I'm going to explore the 'bow out' notion next but I don't see that it applies. detail: The ring group lua eventually calls continue_on_fail=true hangup_after_bridge=true bridge loopback/1234 then, internal dialplan at 1234 does bypass_media=true originate_continue_on_timeout=true hangup_after_bridge=false continue_on_fail=true bridge sofia/internal/5551212(a)voip2pstn.com will always fails with 'Normal temporary failure' and then fall through to voicemail. If bypass_media=true -> bypass_media=false both cases work normally. Using FreeSWITCH version: 1.0.head (git-7f17ad7 2011-10-01 14-05-14 +0200) P.S. Thanks for the pointer to 'jira', I filed the previous issue there. I'm new enough I'll wait to file this until I get some sense of whether it's my mistake somehow before filing one about this. -- Harry Coin Quiet Fountain LLC 2118 Lundy Ln Bettendorf, Iowa 52722 USA Office: 563 332 4970x101 Fax: 563 332 9633 Mobile: 563 650 7800

3 2

Exporting variables through mod_fifo/mod_callcenter & codec issue in mod_fifo
by Marcin Gozdalik 25 Feb '12

25 Feb '12

Hi I am trying to export some variables to consumers (mod_fifo) or callback agents (mod_callcenter). The basic scenario I am trying to achieve is to associate some data with path taken in IVR (in form of exported variable) and pass information on which option was chosen in IVR to the agent. I took a look at the source to find out how mod_fifo/mod_callcenter originate calls to consumer/callback agents and it seems the calls are not associated with A-leg (somebody calling to IVR and transferred to queue). Is it on purpose? Is there a workaround for that? I also spotted strange behavior of mod_fifo when originating calls to consumers. It sets absolute_codec_string for calls originated to consumers to variable_sip_use_codec_name from A-leg, adding PCMA/PCMU if not already present. I find the logic there quite puzzling. If somebody is calling to queue using e.g. G.729, mod_fifo will restrict codecs available in leg to consumer to only G.729, PCMA, PCMU. Why is this narrowing of choice of codecs implemented? Best regards -- Marcin Gozdalik

1 1

Recommend an IP Phone for use behind NAT
by Alan Schmitz 25 Feb '12

25 Feb '12

I am looking into changing my VOIP server to FreeSwitch. My current system uses Polycom SoundPoint IP 335 phones. The server has a public IP, but several of the phones are behind a variety of DSL routers that do NAT. These remote users have reported intermittent problems with one way audio. What IP desk phone would you recommend for a remote user behind a NAT router with a dynamic IP? I'm looking for something of comparable audio and build quality to the Polycoms I currently have. Thanks, Alan

4 3

RECOVERY_ON_TIMER_EXPIRE
by Ira Tessler 24 Feb '12

24 Feb '12

Can someone tell me what a hang up cause of RECOVERY_ON_TIMER_EXPIRE means? I have a customer complaining of dropped calls, and this seems to be the hang up cause. Thanks, Ira Tessler ConnectMe (732) 490-9007 x2 www.connectmevoice.com

4 3

Forcing ptime for A Leg and B Leg call
by Arafath-uz-zaman khan 24 Feb '12

24 Feb '12

Hello Can anyone guide me on following Client----120ms----->FS----20ms---->GW Client<----50ms------FS<----20ms----GW Sip Client will use G729 codec with 120 ptime, Freeswitch will send the call to GW using G279@20 ptime. Freeswitch will send receive call from RTP with G729@20 ptime and will send it to Sip Client using G279@50ptime. Sip Client will be registered at FS using internal SIP Profile. I will use Sangoma transcoding card for it. Can anyone help me to tell the way what i have to do in sip profile and dialplan. With Regards -- Arafath-uz-zaman khan

1 0

Wiki article: XML User Directory Guide and "alphanumeric to numeric user mapping"
by Anton Kvashenkin 24 Feb '12

24 Feb '12

I think that there is a mistake at http://wiki.freeswitch.org/index.php?title=XML_User_Directory_Guide&diff=26… revision. It should be <action application="bridge" data="user/${user_data(${destination_number}@${domain_name} attr id)}@${domain_name}"/> and NOT <action application="bridge" data="user/${user_data(${destination_number}@${domain_name} attr id)}"/> i.e. with @${domain_name} freeswitch@internal> user_data 1002(a)XXX.XXX.XXX.XXX attr id 1002 freeswitch@internal> user_data 1000(a)XXX.XXX.XXX.XXX attr id jugatsu freeswitch@internal> <user id="jugatsu" number-alias="1000"> <user id="1002">

3 4

Speech Phrases
by Brian Foster 24 Feb '12

24 Feb '12

I'm trying to figure out how phrases works in freeswitch, and I have a set of my own voices to use instead of the default ones. I've read http://wiki.freeswitch.org/wiki/Speech_Phrase_Management but I guess I'm not understanding correctly. Situation (might be easier to understand this way): Basically I have a set of prompts (they are actually in my voice) that I plan on using to read off addresses and such from a SIP message. Think "automated dispatching" for fire departments. A script might be: (fire tones) "Indianapolis Engine 123, Indianapolis Medic 123, 928 West Banta Road, Vehicle Fire" x2 "Coordinates 3700 North, 1200 East, Assigned to Marion Ops 1" ...so that information would be set in some variables, like responding_units, address, incident_type, coordinates_1, coordinates_1_direction, coordinates_2, coordinates_2_direction, etc... So the million dollar question is, how do I (temporarily) get freeswitch to use my voice prompts instead of the default prompts when using the say command in the dialplan? Also, how should I organize those prompts so that freeswitch can find them? -- Brian D. Foster Endigo Computer LLC Email: bdfoster(a)endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.

3 9

Brian West quick update
by Ken Rice 24 Feb '12

24 Feb '12

Hey guys, wanted to pass on some info about Brian aka BKW He's still sitting in the hospital in Tulsa, tomorrow, he will go under the knife of a surgeon to remove the offending cyst. so here we are going on another week with Brian away... He'll probably be there are another week... (Docs are saying 5 to 7 days after the surgery before he goes home) Keep i mind the VoiceNetworks offer, I believe he's not even 1/2 way there yet. so everyone fire up those paypal accounts and hit him up... if you wish to send him something in another way, see the Jira Ticket BKW-1 via http://jira.freeswitch.org/browse/bkw-1 all the good info is there He is rapidly running out of sick leave and it looks like they are going to put him on Leave of Absence from work... this means his salary stops... so lets help him out guys K

7 9

xmlcdrd start
by Miha Zoubek 24 Feb '12

24 Feb '12

Hi, I have installed modul xmlcdrd (http://wiki.freeswitch.org/wiki/Xmlcdrd)with Nazim help. Can some one please help me how to stat this modul, what I must do after installed and plugin configuration? Thanks! miha

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