May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Codec Negotiation Error Lync/UCMA -> external provider
by JohnGalt1717 21 May '13

21 May '13

Hi there, I'm a newbee, but I've been reading everything I can and I can't find anything that assists with this. I have a UCMA app that is internal that communicates on the internal sip port (set to 5042) and it bridges with dialplan to the external sip trunk (trying 2, one of which is siproutes). If I enable proxy, the call rings out and I get dead air on the other side, but it won't ever properly establish the call in the UCMA app because of some issue with codecs from what I can tell. Obviously I don't want to use proxy mode, what I really want to do is use it straight up and have FS do the transcoding etc. The problem is that once I do this I get a Codec Negotiation error and I can't figure out why. Anyone have any ideas as to why it's freaking out like this? (log below) Thanks! 2013-05-21 12:09:41.570884 [DEBUG] sofia_glue.c:1219 Local SDP: v=0 o=FreeSWITCH 1369129003 1369129004 IN IP4 10.8.0.158 s=FreeSWITCH c=IN IP4 10.8.0.158 t=0 0 m=audio 23578 RTP/AVP 9 0 8 101 13 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-05-21 12:09:41.570884 [DEBUG] mod_sofia.c:114 (sofia/external/+16032897972) State Change CS_INIT -> CS_ROUTING 2013-05-21 12:09:41.570884 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/+16032897972 [BREAK] 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+16032897972) State INIT going to sleep 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+16032897972) Running State Change CS_ROUTING 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+16032897972) State ROUTING 2013-05-21 12:09:41.570884 [DEBUG] mod_sofia.c:137 sofia/external/+16032897972 SOFIA ROUTING 2013-05-21 12:09:41.570884 [DEBUG] switch_ivr_originate.c:67 (sofia/external/+16032897972) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-05-21 12:09:41.570884 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/+16032897972 [BREAK] 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+16032897972) State ROUTING going to sleep 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+16032897972) Running State Change CS_CONSUME_MEDIA 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:489 (sofia/external/+16032897972) State CONSUME_MEDIA 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:489 (sofia/external/+16032897972) State CONSUME_MEDIA going to sleep 2013-05-21 12:09:41.570884 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/+16032897972 [BREAK] 2013-05-21 12:09:41.570884 [DEBUG] sofia.c:5742 Channel sofia/external/+16032897972 entering state [calling][0] 2013-05-21 12:09:43.891308 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/+16032897972 [BREAK] 2013-05-21 12:09:43.891308 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/+16032897972 [BREAK] 2013-05-21 12:09:43.891308 [INFO] sofia.c:966 sofia/external/+16032897972 Update Callee ID to "+16032897972" <sip:+16032897972@4.30.136.195> 2013-05-21 12:09:43.891308 [DEBUG] sofia.c:5742 Channel sofia/external/+16032897972 entering state [proceeding][183] 2013-05-21 12:09:43.891308 [DEBUG] sofia.c:5751 Remote SDP: v=0 o=Sonus_UAC 10608 14579 IN IP4 207.239.33.59 s=SIP Media Capabilities c=IN IP4 207.239.33.59 t=0 0 m=audio 10460 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=maxptime:20 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:2714 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:2714 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:1813 Set Codec sofia/external/+16032897972 PCMU/8000 20 ms 160 samples 64000 bits 2013-05-21 12:09:43.891308 [DEBUG] switch_core_codec.c:111 sofia/external/+16032897972 Original read codec set to PCMU:0 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:2876 Set 2833 dtmf send payload to 101 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:3800 AUDIO RTP [sofia/external/+16032897972] 10.8.0.158 port 23578 -> 207.239.33.59 port 10460 codec: 0 m s: 20 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:2659 Starting timer [soft] 160 bytes per 20ms 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: START SESSION INITIALIZATION. sID=4. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: ZID=306130383030396530393363. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Loading User's profile: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: allowclear: OFF 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: autosecure: ON 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: disclose_bit: OFF 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: signal. role: Initiator 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: TTL: 4294967295 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: SAS schemes: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 B256 2013-05-21 12:09: 43.891308 [DEBUG] switch_rtp.c:1231 B32 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Ciphers: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 AES3 2013-05-21 12:09:4 3.891308 [DEBUG] switch_rtp.c:1231 AES1 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: PK schemes: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 EC25 2013-05-21 12:09: 43.891308 [DEBUG] switch_rtp.c:1231 DH3k 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 DH2k 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 Mult 201 3-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: ATL: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 HS32 2013-05-21 12:09: 43.891308 [DEBUG] switch_rtp.c:1231 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Hashes: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 S256 2013-05-21 12:09:4 3.891308 [DEBUG] switch_rtp.c:1231 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Session initialization - DONE. sID=4. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: ATTACH NEW STREAM to sID=4: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp]: Stream ID=0 UNKNOWN switching <NONE> ---> <ACTIVE>. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Empty slot was found - initializing new stream with ID=4. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Preparing ZRTP Hello according to the Session profile. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: ATTACH NEW STREAM - DONE. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp engine]: START STREAM ID=4 mode=CLEAR state=ACTIVE. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp]: Stream ID=4 CLEAR switching <ACTIVE> ---> <START>. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp utils]: Send <HELLO> ssrc=1428962591 seq=54611 size=144. Stream 4:CLEAR:START 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:4145 Set 2833 dtmf send payload to 101 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:4151 Set 2833 dtmf receive payload to 101 2013-05-21 12:09:43.891308 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/+16032897972! 2013-05-21 12:09:43.891308 [DEBUG] switch_channel.c:3273 Send signal sofia/internal/+16032897972(a)173.255.174.20 [BREAK] 2013-05-21 12:09:43.891308 [DEBUG] switch_channel.c:3277 (sofia/external/+16032897972) Callstate Change DOWN -> EARLY 2013-05-21 12:09:43.911297 [DEBUG] switch_ivr_originate.c:411 Setting codec string on sofia/internal/+16032897972(a)173.255.174.20 to PCMU@8000h@20i 2013-05-21 12:09:43.911297 [INFO] switch_ivr_originate.c:3416 Sending early media 2013-05-21 12:09:43.911297 [ERR] switch_core_media.c:2549 a=crypto in RTP/AVP, refer to rfc3711 2013-05-21 12:09:43.911297 [ERR] mod_sofia.c:2081 CODEC NEGOTIATION ERROR. SDP: v=0 o=- 1 0 IN IP4 10.8.0.158 s=session c=IN IP4 10.8.0.158 b=CT:10000 t=0 0 m=audio 2908 RTP/AVP 9 112 111 0 8 116 13 118 97 101 c=IN IP4 10.8.0.158 a=rtpmap:9 G722/8000 a=rtpmap:112 G7221/16000 a=fmtp:112 bitrate=24000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:116 AAL2-G726-32/8000 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,36 a=rtcp:2909 a=label:main-audio a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BqtEfRv/sMasd3GS0I4FJRX5hh7BKVt+r8/vKbOk -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Codec-Negotiation-Error-Lync-… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

DTMF
by Steven Schoch 21 May '13

21 May '13

One of my users complains of many problems calling automated systems that take DTMF input. In this case it was an IRS payment entry system that would skip or duplicate button presses. Our setup is pretty generic: Polycom SoundPoint IP 320 SIP -> dedicated LAN -> Freeswitch (on CentOS, on XenServer) -> Comcast static IP -> Flowroute.com -> some IRS 800 number. Is this just the way it is, or is there something I can do to make it work better? -- Steve

2 3

files.freeswitch.org down
by Cesar Bermudez 21 May '13

21 May '13

hi all, is the domain files.freeswitch.org down? i cannot download the finish the install ( the sounds ) or its just me? Regards.

3 3

answer calls during opening message of fifo
by Jason Holden 21 May '13

21 May '13

I would lke a welcome message to play when a caller connects to FIFO where the agents are able to answer their phone while this message is played. Currently I am using play sound but it does not ring the agent phones during this messages since fifo has been entered. Does anyone have any recommendations?

1 0

mod_unimrcp dont send invite while using a Mrcp v2 profile
by Leonardo P. Bidinoto 21 May '13

21 May '13

Hi All, Im new at using the mod_unimrcp. I'm having some troubles when configuring to using several profiles at same time. I have configured 3 profiles(2 with Nuance and 1 Verbio, all using MRCPv2). But when i try to use a TTS or ASR from any of those profile, FS channel get stuck and i get a warning message telling that the time for the MRCP server to respond have expired. I have used wireshark to capture the packages, and when i call speak or detect_speech function of FS, i dont see any SIP INVITE going from FS to the MRCP Server. Did i miss something at configuring profiles? Did anyone get something similar to this situation? I'm using the examples from FS mod_unimrcp page. Any tip will help a lot. Thanks -- Leonardo Pires Bidinoto Voice Technology www.voicetechnology.com.br

2 1

GXP1405 G729 audio problem
by Juan Antonio Ibañez Santorum 21 May '13

21 May '13

Hello, Anyone had problems using G279 codec with Grandstream GXP1405. GXP1405 is connected to FS which is connected to an Asterisk box. FS has no codec installed so it is in passthrough mode. The audio received in the phone is bad. Sometimes volume goes down and it has low quality. Working connected directly to Asterisk using G729 box goes well. Working through FS using G711 goes well. If I analyze both flows, from Asterisk and from FS, inside FS box using G729, all audio is OK. So I don't know why phone plays it bad... Any idea? Regards

1 0

Re: [Freeswitch-users] custom sip header not working in dialplan
by windy 21 May '13

21 May '13

Hi, Freeswitch Team There is problem for to modify the outbound sip header. As the freeswitch wiki says, it can modify/add custom sip header via below method: <action application="set" data="sip_h_X-Charge=1"/> <action application="set" data="sip_rh_X-Reason=Destination Number Not in Footprint"/> <action application="set" data="sip_bye_h_X-Accounting=Some Accounting Data"/> Now, I add the red lines in dialplan/default.xml. But only the custom header prefixed with "sip_h_" can be shown in the SIP INVITE message. Other custom headers prefixed with "sip_bye_h" and "sip_ph"... not found in the related messages(BYE, 183). Could you kindly give comments about this case? <context name="from-pstn-to-sip"> <extension name="to-sip">  <condition field="destination_number" expression="^(.{1,})$" break="never"> <action application="set" data="destnumber=$1"/> <anti-action application="set" data="destnumber=unknown"/> </condition>  <condition field="destination_number" expression="^(.*)$"> <action application="set" data="hangup_after_bridge=yes" /> <action application="set" data="tone_detect_hits=1" /> <action application="export" data="fax_enable_t38_request=true" /> <action application="export" data="fax_enable_t38=true" /> <action application="tone_detect" data="faxdisable 1100 r +5000 disable_ec 1"/> <action application="export" data="execute_on_answer=tone_detect fax_disable_ec 2100 r +5000 t38_gateway 'self nocng'" /> <action application="set" data="sip_contact_user_replacement=${destnumber}"/> <action application="set" data="hangup_after_bridge=yes"/>  <action application="export" data="sip_info_when_no_2833=false" /> <action application="set" data="sip_h_X-Charge=1"/> <action application="set" data="sip_rh_X-Reason=Destination Number Not in Footprint"/> <action application="set" data="sip_bye_h_X-Accounting=Some Accounting Data"/>    <action application="bridge" data="sofia/external/sip:${destination_number}@172.20.2.17:5062"/> <action application="hangup" data="${originate_disposition}"/> </condition> </extension> Thanks Windy

1 0

Problem with enable-post-var
by Anatolii 21 May '13

21 May '13

Hello all. In mod_xml_curl in binding configuration I try make dynamic "gateway-url". For example, <param name="gateway-url" value="http://hostname/configuration/<b>$${my_var}*" bindings="configuration"/> Also try to add 'enable-post-var' -- didn't work too. Please, help anyone. Thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-enable-post-var-… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 1

UUID - how unique is it?
by Telecube - John 21 May '13

21 May '13

How unique is uuid in the cdr records? Coming from asterisk where uid is about as unique as a black bowling ball.... Can I count on uuid to be totally unique in my billing records? -- Regards, John Matich Telecube Pty Ltd www.telecube.com.au <http://www.telecube.com.au> Ph: 13CUBE (132823)

10 13

FreeSWITCH RTMFP support
by Adam Raszynski 21 May '13

21 May '13

Hi Are there any plans to add support form Adobe RTMFP protocol? I know that there is mod_rtmp, but it supports only RTMP which is TCP-based. RTMFP is UDP based and designed specially for P2P communications and should give much smaller delays Kind Regards

2 1