October 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

issue of cdr_mongodb
by erhuabushuo 24 Oct '13

24 Oct '13

established connection, after finished call, when it try to insert cdr to mongodb, I got this error '2013-10-23 17:21:29.745772 [ERR] mod_cdr_mongodb.c:393 mongo_insert: (error code 15)'

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mod_shout in Windows distribution?
by Rafal Gwizdala 24 Oct '13

24 Oct '13

Hi, is there a reason why mod_shout is not included with Windows binary distribution? Best regards RG

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Parsing from sip header
by Miha 24 Oct '13

24 Oct '13

Hi, I am sending to FS some attribute in header which I have added on proxy side. How can I parse this header, to get this data and send it further? I can not see it in varibles which are printed with info application. I added: Moved: 38618108751. BR, Miha

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Re: [Freeswitch-users] codec transcodation not working
by neeraj.p 24 Oct '13

24 Oct '13

Hey Steven Here is the result of command - sofia status profile internal I can see speex in the list. freeswitch@127.0.0.1:5222@internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 172.16.138.74,172.16.138.74 Dialplan XML Context public Challenge Realm auto_from RTP-IP 172.16.138.74 SIP-IP 172.16.138.74 URL sip:mod_sofia@172.16.138.74:5060 BIND-URL sip:mod_sofia@172.16.138.74:5060;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMU,GSM,PCMA,speex@8000h@20i CODECS OUT PCMU,GSM,PCMA,speex@8000h@20i TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 1 --Neeraj

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Individual party recording
by Siri MM 24 Oct '13

24 Oct '13

Hi, Is it possible for me to record each party of a call individually? If I were to transfer the call a few times to different extensions, I would like store each party's conversation separately - is this feasible? Thanks!

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loopback_bowout
by Peter Steinbach 24 Oct '13

24 Oct '13

Hello, I have a strange behaviour with forwarding a call from an external gateway to another number on the same gateway. * when I forward a call from an internal number to an external number via loopback, I firstly get 4 channels. After answering the call, I have 2 channels remaining and audio is fine * when I forward a call from an*_*external_ number to an external number via loopback, I firstly get 4 channels. After answering the call, I still have 4 channels remaining and audio is _not_ there. Freeswitch is dialling via "external" profile via port 5080. I also tried to set "loopback_bowout=true", but this did not change anything. The dialstring is: <action data="{sip_from_uri=0610xxxxxxx@sip.domain.de}sofia/gateway/SIP-68xxxxx/017xxxxx@sip.domain.de" application="bridge" /> The dialplan for both scenarios is the same. Anybody had the same issue and knows how to overcome this? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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codec transcodation not working
by neeraj.p 24 Oct '13

24 Oct '13

Hey, I tried to establish audio call between two clients having different codec supports . I expected that transcoding would take place. But I am getting a codec negotiation error . Here is the information about client I am using leg A : suppoerted codec : speex leg B: supported codec : G722,PCMU,PCMA,GSM,OPUS Here is some lines from my freeswitch logs 2013-10-22 10:31:12.795786 [INFO] switch_ivr_originate.c:1190 Sending early media 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[G722:9:8000:20:64000] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMU:0:8000:20:64000] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMA:8:8000:20:64000] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[GSM:3:8000:20:13200] 2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:3064 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-22 10:31:12.795786 [ERR] mod_sofia.c:2122 CODEC NEGOTIATION ERROR. SDP: v=0 o=Zoiper 0 0 IN IP4 120.63.38.94 s=Zoiper c=IN IP4 120.63.38.94 t=0 0 m=audio 11525 RTP/AVP 110 a=rtpmap:110 speex/8000 Here is some info about my freeswitch version - 1.4 beta codecs supported - codec,ADPCM (IMA),mod_spandsp codec,AMR,mod_amr codec,B64 (STANDARD),mod_b64 codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_spandsp codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_spandsp codec,G.726 16k (AAL2),mod_spandsp codec,G.726 24k,mod_spandsp codec,G.726 24k (AAL2),mod_spandsp codec,G.726 32k,mod_spandsp codec,G.726 32k (AAL2),mod_spandsp codec,G.726 40k,mod_spandsp codec,G.726 40k (AAL2),mod_spandsp codec,G.729,mod_g729 codec,GSM,mod_spandsp codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_spandsp codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex codec,VP8 Video (passthru),mod_vp8 My sip_profile configurations <param name="disable-transcoding" value="false"/> <param name="inbound-late-negotiation" value="true"/> I also tried with inbound-late-negotiaion=false . But still getting codec negotiation error. Please help . Regards, Neeraj

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Out of band DTMF in Javascript
by Murthy Gandikota 24 Oct '13

24 Oct '13

Hi Can anyone please confirm that the following template is RFC2833 compliant out-of-band DTMF for the #3#? "%(200,0,941.0,1477.0);%(200,0,697.0, 1477.0);%(200,0,941.0,1477.0)"; I am trying to use TeleTone to generate the #3# response to a caller as follows: var pound3pound ="v=-7;%(200,0,941.0,1477.0);%(200,0,697.0, 1477.0);%(200,0,941.0,1477.0)"; var tts = new TeleTone(session); tts.generate(pound3pound); I can hear the digits play, but don't know if they are RFC2833 compliant out-of-band DTMF. Thanks Murthy

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disconnect after 30-60 with freeswitch and polycom phones on same network
by Allen Schultz 23 Oct '13

23 Oct '13

I'm having an issue with the 30-60 second disconnect on both calls (from outside sip provider) and local voicemail (seperate incident, but related). Below I have some pastebin logs of some information with some preliminary steps with the #freeswitch irc group. Event: with sofia siptrace on <http://pastebin.com/cvhjAJpE> sofia status <http://pastebin.com/C6uFZTTC> sofia status profile internal <http://pastebin.com/Km059vhC> sofia status profile external <http://pastebin.com/eVwZTNJy> Polycom soundpoint ip 430 rev a configured polycoms to go directly to 192.168.123.6 Any help will be appreciated. Allen

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Audio quality issues
by Grant Bagdasarian 23 Oct '13

23 Oct '13

Hello, I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this. We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue's. Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren't doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test. These numbers are for one way audio, where Sipp doesn't echo the RTP back. These numbers get lower once Sipp echo's the RTP. I've tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn't that much. I disabled all the modules we don't need, like CDR's, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant

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