June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

What's the best language for implementing an IVR on Freeswitch
by Terry Ray 07 Jun '13

07 Jun '13

Greetings, While I've been very pleased with the stability and performance of FreeSWITCH itself, I'm having to scrap the Ruby implementation I was using to drive the call menu functionality for the IVR due to it's inability to maintain connectivity with FS via the inbound event socket. Would love some suggestions on the best approach. Lua? or some other approach? Mainly I'm looking for something that will be very stable at higher call volumes. Best regards, Terry

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mod_voicemail + send_silence_when_idle
by Yuriy Nasida 06 Jun '13

06 Jun '13

Hi guys, One of my carriers drops a call if FS deos not send RTP media (10-20 sec). Thus issues happens, when voicemail module asks to leave a message. I tried to use send_silence_when_idle. I hear a noise from FS but not in the moment of recording... Please advice,Thank.

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cannot make call, error INCOMPATIBLE_DESTINATION
by Khue Nguyen Minh 06 Jun '13

06 Jun '13

Hi All, I'm using FS to receive call forward from GMSC. But, I have error INCOMPATIBLE_DESTINATION when new incoming call. This is log from fs_cli, please solve for me. Audio Codec Compare [PCMA:8:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[GSM:3:8000:20:13200] Audio Codec Compare [AMR-WB:101:8000:5:0]/[G722:9:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMU:0:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMA:8:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[GSM:3:8000:20:13200] Audio Codec Compare [PCMU:0:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[GSM:3:8000:20:13200] Substituting codec PCMU@20i@8000h No audio codec available No 2833 in SDP. Disable 2833 dtmf and switch to INFO Send signal sofia/internal/963728839(a)x.x.x.x [BREAK] (sofia/internal/963728839(a)x.x.x.x) Callstate Change RINGING -> HANGUP Hangup sofia/internal/963728839(a)x.x.x.x [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Send signal sofia/internal/963728839(a)x.x.x.x [KILL] Send signal sofia/internal/963728839(a)x.x.x.x [BREAK] sofia/internal/963728839(a)x.x.x.x skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) (sofia/internal/963728839(a)x.x.x.) State EXECUTE going to sleep (sofia/internal/963728839(a)x.x.x.x) Running State Change CS_HANGUP (sofia/internal/963728839(a)x.x.x.x) State HANGUP Channel sofia/internal/963728839(a)x.x.x.x hanging up, cause: INCOMPATIBLE_DESTINATION Responding to INVITE with: 488 sofia/internal/963728839(a)x.x.x.x Standard HANGUP, cause: INCOMPATIBLE_DESTINATION Brs, Khue Nguyen.

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Ring an extension on incoming calls
by Sherwood Hu 06 Jun '13

06 Jun '13

Hi all, We are considering to replace current PBX with freeswitch. I am new to this topic - and find that freeswitch is fantastic. Now I have a question - I setup the freeswitch to transfer to IVR extension when the main line number is called. Our current PBX first rings an extension (to allow the caller information displayed on the phone) then plays the IVR. I want this functionality in freeswitch too. How can I do so? Thanks in advance.

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why disable_hold is not working?
by Vincent Xia 06 Jun '13

06 Jun '13

why disable_hold is not working? i set <action application="set" data="disable_hold=true"/> in the dialplan and also <param name="disable_hold" value="true"/> in sip profiles, nothing happened. another question is, disable_hold is described as a channel variable, i wonder how FS affects the channel by setting <action application="set" data="disable_hold=true"/> in the dialplan? for example, <extension name="disable_hold"> <condition field="destination_number" expression="^(11[01][0-9])$"> <action application="answer"/> <action application="set" data="disable_hold=true"/> </condition> </extension> in this extension, which one, the calling party or called party, will be disabled?

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Freeswitch SDP manipulation with dynamic payload types
by Ashwin Rath 06 Jun '13

06 Jun '13

Hello all Here is our ingress SDP to freeSWITCH (partial text only) m=audio 40000 RTP/AVP 97 96 98 a=rtpmap:97 G.729b/8000 a=rtpmap:96 G.729ab/8000 a=rtpmap:98 G.729a/8000 and this is the egress from FS m=audio 20788 RTP/AVP 18 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Notice that FS is substituting the G729 flavors with just 18 in the outgoing SDP. Can we have the same codecs offered in the outgoing SDP as well ? --Ashwin

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Multi network interface
by Khue Nguyen Minh 06 Jun '13

06 Jun '13

Hi all, I run FreeSWITCH on a server have two network interfaces. But, I see that, FS only listen/bind to one interface. I am using configure default of FS. How I can make FS listen/bind on two interfaces? Brs, Khue Nguyen.

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How to get some RTP stats
by Mino Haluz 05 Jun '13

05 Jun '13

Hi, is there any way how to track: 1) packet loss 2) jitter 3) some other interesting RTP statistics in FS ? Thanks Mino

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RECURSIVE ERROR
by Rodrigo Pinto 05 Jun '13

05 Jun '13

Hello, We recently migrated to latest stable version 1.2.9. And in the console this error is appearing randomly: 2013-06-05 18:14:00.089079 [ERR] switch_ivr.c:144 RECURSION ERROR! It's not the best idea to call things that collect input recursively from an input callback. I see that this version has this check in ./include/switch_types.h: #define MAX_ARG_RECURSION 25 #define arg_recursion_check_start(_args) if (_args) { \ if (_args->loops >= MAX_ARG_RECURSION) { \ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, \ "RECURSION ERROR! It's not the best idea to call things that collect input recursively from an input callback.\n"); \ return SWITCH_STATUS_GENERR; \ } else {_args->loops++;} \ } Any idea how to easily increase MAX_ARG_RECURSION or disable it like older versions? Thanks!

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FS Choking on a 200 Response w/ a Duplicate "From" Header
by JP 05 Jun '13

05 Jun '13

I thought I would bounce this off the mailing list before filing a Jira ticket. I'm facing an issue where FS doesn't acknowledge a 200 OK response with a duplicate "From" header: recv 422 bytes from udp/[10.191.210.149]:5060 at 17:21:22.634270: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.191.210.147:5080;rport;branch=z9hG4bKNv1g56Djj6BKj From: "" <sip:0000000000@10.191.210.147>;tag=By8FpSr2m6NvF From: "" <sip:0000000000@10.191.210.147>;tag=By8FpSr2m6NvF To: <sip:1000@10.191.210.149:5060>;tag=1 Call-ID: 1efaef87-48a7-1231-d08a-000c29ccde0a CSeq: 44890363 INVITE Contact: <sip:10.191.210.149:5060;transport=UDP> Content-Type: application/sdp Content-Length: 0 Although I've not yet stepped through the code, it is my suspicion that the Sofia SIP stack is not accepting the response. Aside from the re-transmissions of the response that I'm seeing in the logs, there are no other error, info, or trace statements coming from any other FS module. RFC 3261 seems a bit vague in my opinion as to whether or not this is even allowed. As a side note, this issue came up while doing integration testing with OpenSER 1.3.4. OpenSER seems to stick in the duplicate From header whenever it does a serial proxy to multiple carriers and the first carrier leg times out. I'm wondering if anyone on the mailing list was familiar with this issue and perhaps knew of some fix/workaround on the FreeSWITCh end. Thanks, JP

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