- FreeSWITCH-users - FreeSWITCH Mailing Lists

One Way Audio
by Sean Devoy 22 May '13

22 May '13

Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean

3 9

mod_cdr_csv is not working!
by Shahzad Bhatti 22 May '13

22 May '13

Hi, after loading *mod_cdr_csv, *when i originate test calls the cdr is not appended in Master.csv file most of the time but sometime cdr is appended in the Master.csv that show that mod_cdr_csv is configured. Also when i try to use *cdr_csv rotate* command on fs_cli it also not work. even i also try to do using the perl script example available on following url https://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_Perl_Script_for_CDR_in… i want to know why this is happening on my server and how i can fix these issues 1. cdrs are not updated in the csv file; 2. rotate command is not working; any reply is highly appreciated; Regards Shahzad Bhatti

2 1

Routing on the basis of RURI
by Lucas Tehbing 22 May '13

22 May '13

Dear Guys, I have been using Opensips as SBC but now want to move to Freeswitch because of Jitter buffer module and many other interesting features. My Call flow is .... SIP UA (Behind NAT) -------------> SBC (Currently opensips)----->Sip Provider (Configured at SIP UA) When SIP message is received at SBC, SBC checks if RURI belongs to some other domain (if(uri==myself) ) if yes it reads the RURI and forwards (State fully) the message (Register, Invite , BYE all kinds of messages). It also modifies SDP in order to be in media path. I want to implement FS in same scenario. The issue is that I cannot configure RURIs on FS, Freeswitch should read from received Request where to forward the message. I am still in the process to learn Freeswitch. If anyone can point me to right resource it will save lot of time. Regards, Lucas

1 1

Switch on/off call recording
by Ashish gautam 22 May '13

22 May '13

Hi, I have setup a little call center application using mod_callcenter. Now what I want is to record all the calls based on the click of an ON?OFF button on the web app. Is there any variable or something like that which I can set like True or False to switch the call recording on or off? Also I want to know if the recording is possible per agent or not i.e. the call session to be recorded for a particular agent. Please throw some light on this. Thanks in advance. Regards -Ashish

2 3

Notification of an event in the dialplan
by Christopher Aloi 21 May '13

21 May '13

I'm currently using FreeSwitch between two systems as a "rate limiter' with the limit application. I bring calls in, run through a hash to identify a loop - if my hash variable is met, I transfer the call to a new extension where I log the event and return a 503 to the source. If not I return a 302 to the source to send the call on its way. I'd like a way to identify these events to our NOC via email or a SNMP trap but I am not sure the best way to do it. The alert would fall into the 503 extension. What is the best way to accomplish this? I'd like to keep the call flow as simple as possible. Any ideas? Thanks - - Chris

4 5

Separating Signaling and Media Among 2 FreeSWITCHs
by JP 21 May '13

21 May '13

Is there any way to separate the signaling and media for a call across 2 FreeSWITCH servers? Suppose that I would ideally like to transfer a call from one FS server to another, but the user agent doesn't support any kind of SIP methods to accomplish this (ex. REFER, INVITE with "Replaces" header, etc.). Is there a way to re-route the media to another FS server while keeping the signaling anchored on the original FS? Thanks JP

2 2

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