July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem with rtp streams and stair-stepped (or ramped) increased lag
by Nathan Neulinger 10 Jul '13

10 Jul '13

Having some occasional user reports of problem calls where they are fine, and then something happens on the call and lag increases significantly, sometimes up to 2+ seconds. I've got severe examples on both cisco and polycom phones. What I'm seeing looks like the call is going perfectly for a period of time (sometimes up to a few minutes), and then there is a period of time with higher latency packets, and the overall delay in the graph increases steadily at that point, and never recovers. It sounded like rtp-autoflush-during-bridge might be something relevant, but that's set to the default. Also saw "rtp-rewrite-timestamps", but unclear if that is necessary here. I've got pcap captures of the rtp streams if that'll help any. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

8 34

Jira and Developers requesting more information
by Ken Rice 10 Jul '13

10 Jul '13

Greeting FreeSWITCHers! So, I hate to do this, but we¹ve got the perfect example here of a ticket in Jira... http://jira.freeswitch.org/browse/FS-4918 Last December, the reporter gives us a fairly good bug report with most of the good information, but also Anthony was able to deduce a possible problem, and present a patch. Several days later the reporter comes back that the patch solves a segfault issue, but the developer requested more information as he added some debugging into the patch to issolate the root cause... This tells me this patch probably fixes the issue, but theres a possibility that the problem may still be lurking int he shadows for later... Now here we are 5 months later and the reporter hasn¹t responded with the requested info and we are left with nothing to do but close the ticket knowing he hasn¹t poped back up complaining about the segfault and nothing more. Now, you may be asking whats the problem here, the problem has been fixed? The problem comes from the fact as part of his commitment to supporting FreeSWITCH, Anthony (the developer) committed a fair bit of his time to track down and fix the issue for which he asked in trade nothing but more information... Which is well with in the spirit of Open Source Software... But the reporter kinda left us hanging here. He got a developer to work on (and probably fix) his problem but doesn¹t pay the developer back for his work with the needed requested... And now we have another open stale bug waiting around on the reporter who will probably never submit the info... So please if the developer ask for more info give it to them, that¹s what are you are expected to do in exchange for their time at the very least. TL;DR version... If developers ask for more info, don¹t be a tool, give them the requested info... -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

1 0

Re: [Freeswitch-users] FreeSWITCH Start up Script - Ubuntu Server
by sreekanth 10 Jul '13

10 Jul '13

just add "/usr/local/freeswitch/bin/freeswitch -nc" to /etc/rc/local before exit 0 My rc.local file looks like below #!/bin/sh -e # # rc.local # # This script is executed at the end of each multiuser runlevel. # Make sure that the script will "exit 0" on success or any other # value on error. # # In order to enable or disable this script just change the execution # bits. # # By default this script does nothing. /usr/local/freeswitch/bin/freeswitch -nc exit 0 ------------ cheers ! http://askadmin.com http://sipcosystems.com -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-Start-up-Script-Ub… Sent from the freeswitch-users mailing list archive at Nabble.com.

6 7

mod_callcenter and start_dtmf
by Mimiko 10 Jul '13

10 Jul '13

Hello. Recently I have found problem which was in callcenter application. Agents randomly had heard a beep sound like calling members where pushing some number to send a dtmf done. This was annoying, because this was in a middle of conversation. This tone had appearing randomly not on every incoming call to callcenter. It was high volume of 0.5-1 sec long. I couldn't track the source of this sound, as in the resulting recorded file of the conversation there where no suspicious sounds, nor a beep from callers, nor the beep from or to agents. So I was lost. Several days ago I remembered about using start_dtmf application on incoming calls from phone provider because the do not use rfc standard for dtmf. More strange thing was that this strange beep sound didn't appear on regular incoming call to some extension. It was only when external callers from phone provider were calling into callcenter number. So I used stop_dtmf just before sending incoming call to callcenter application. With this command those strange beeps disappeared. Then I've changed start_dtmf to spandsp_start_dtmf and didn't use spandsp_stop_dtmf. Beeps does not appear too. So there seems to be a problem with core handling of in-band dtmf with start_dtmf aplication when call is routed into callcenter, and the callcenter bridged the call to the agent. Should I rise an issue to the jira, or this is an internal known issue of core dtmf handling and will not be resolved? Mimiko desu.

1 0

Connecting asterisk to freeswitch
by Ashwin Jain 10 Jul '13

10 Jul '13

Hi, I am trying to connect asterisk to freeswitch. For this, I have added a contact in sip_profiles/users.xml, and asterisk is connecting FS as a sip contact. When I am trying to dial to asterisk contact, asterisk is not getting any SIP packets. Following is the registration information of asterisk contact: Registrations: ================================================================================================= Call-ID: 4e46179c713c03f42d1a6dfe3b104777(a)127.0.0.1 User: 102(a)freeswitch.abcd.co Contact: "user" <sip:102@182.172.122.40> Agent: Asterisk PBX 1.6.2.12 Status: Registered(UDP)(unknown) EXP(2013-07-10 03:09:02) EXPSECS(128) Host: freeswitch IP: 182.172.122.40 Port: 14638 Auth-User: unknown Auth-Realm: freeswitch.abcd.co MWI-Account: 102(a)freeswitch.abcd.co Total items returned: 1 ===================================================================== -- Thanks and Regards, Ashwin Jain

3 2

Firefox WebRTC 'fingerprint' SDP attribute missing 200OK response
by Karl Mack 10 Jul '13

10 Jul '13

All: If possible I would like to inquire why my local build of 1.4.0 B3 does not return a 'fingerprint' SDP attribute in the 200 OK response for a WebRTC origination from Firefox v24.0a1. I completed a similar WebRTC call via Freeswitch website 'https://webrtc.freeswitch.org/sipml5/' and the 'fingerprint' SDP attribute exists. Am I missing an configuration setting? *-- Local Build : freeswitch-1.4.0.beta3 (6/28/2013) --* INVITE sip:9600@example.com SIP/2.0 Call-ID: 1373406010440 CSeq: 1 INVITE From: "Test System" <sip:5275@example.com>;tag=1373406014154 To: <sip:9600@example.com> Via: SIP/2.0/WS B9x0pJyOAiMG.invalid;branch=z9hG4bK1373405991583;rport Max-Forwards: 70 Content-Type: application/sdp User-Agent: TestWebPhone/1.00 Allow: INVITE,ACK,CANCEL,BYE Contact: <sip:5275@B9x0pJyOAiMG.invalid;transport=ws> Content-Length: 588 v=0 o=Mozilla-SIPUA-24.0a1 1327 0 IN IP4 0.0.0.0 s=SIP Call t=0 0 a=ice-ufrag:0dbd1e66 a=ice-pwd:9cc76e3ba9334f2e8369fc5f730d4839 a=fingerprint:sha-256 BE:83:F6:B9:A0:B7:73:0A:26:A0:2A:BC:54:9D:5F:E9:1A:9D:A3:5A:60:66:A7:EF:05:42:31:42:79:B6:C9:BB m=audio 64358 RTP/SAVPF 109 0 8 101 c=IN IP4 192.168.129.83 a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:0 1 UDP 2111832319 192.168.129.83 64358 typ host a=candidate:0 2 UDP 2111832318 192.168.129.83 62653 typ host SIP/2.0 200 OK Via: SIP/2.0/WS B9x0pJyOAiMG.invalid;branch=z9hG4bK1373405991583;rport=64794;received=192.168.129.83 From: "Vail Systems" <sip:5275@vailsys.com>;tag=1373406014154 To: <sip:9600@vailsys.com>;tag=QD139mFva9jNg Call-ID: 1373406010440 CSeq: 1 INVITE Contact: <sip:9600@74.63.156.145:5090;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 762 Remote-Party-ID: "Outbound Call" <sip:9600@vailsys.com>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1373356842 1373356843 IN IP4 74.63.156.145 s=FreeSWITCH c=IN IP4 74.63.156.145 t=0 0 a=msid-semantic: WMS Xusp8lc7KzweB2mads8BzqXOOdfmpEjd m=audio 49172 RTP/SAVPF 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=ptime:20 a=rtcp:49173 IN IP4 74.63.156.145 a=ssrc:3722431166 cname:YBsXNuUnUhrmaDmd a=ssrc:3722431166 msid:Xusp8lc7KzweB2mads8BzqXOOdfmpEjd a0 a=ssrc:3722431166 mslabel:Xusp8lc7KzweB2mads8BzqXOOdfmpEjd a=ssrc:3722431166 label:Xusp8lc7KzweB2mads8BzqXOOdfmpEjda0 a=ice-ufrag:WkEmBs0fv5sLzBOn a=ice-pwd:37bGsVTc3ap3EB6q a=candidate:6683873826 1 udp 659136 74.63.156.145 49172 typ host generation 0 a=candidate:6683873826 2 udp 659134 74.63.156.145 49173 typ host generation 0 *---------------------------------------------------------* *-- O&A https://webrtc.freeswitch.org/sipml5/ -- *INVITE sip:9386@webrtc.freeswitch.org SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKcxkVDkhksN02zm8U31EiPCdwiTReMKQw;rport From: "Karl mack"<sip:1000@webrtc.freeswitch.org>;tag=RtwA31FZToUkxXyeZOLy To: <sip:9386@webrtc.freeswitch.org> Contact: "Karl mack"<sips:1000@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;+sip.ice;language="en,fr" Call-ID: eb87cf95-ea84-ae23-d7ee-309431fce563 CSeq: 21413 INVITE Content-Type: application/sdp Content-Length: 1604 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2013.05.24 Organization: Doubango Telecom v=0 o=Mozilla-SIPUA-24.0a1 2839 1 IN IP4 0.0.0.0 s=Doubango Telecom - firefox t=0 0 a=ice-ufrag:6abd2c98 a=ice-pwd:ee047a97e6b3603180c693e7a065ae16 a=fingerprint:sha-256 B7:71:34:0E:4B:B1:B8:74:7E:02:DD:A0:3D:F4:19:98:DA:BD:B4:D3:85:3A:A6:3E:C2:BE:53:93:7A:17:E7:79 m=audio 51710 UDP/TLS/RTP/SAVPF 109 0 8 101 c=IN IP4 63.210.102.138 a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:0 1 UDP 2111832319 192.168.129.83 63474 typ host a=candidate:1 1 UDP 1692467199 63.210.102.138 51710 typ srflx raddr 192.168.129.83 rport 63474 a=candidate:2 1 UDP 1692467199 63.210.102.138 52129 typ srflx raddr 192.168.129.83 rport 63474 a=candidate:0 2 UDP 2111832318 192.168.129.83 63475 typ host a=candidate:1 2 UDP 1692467198 63.210.102.138 50308 typ srflx raddr 192.168.129.83 rport 63475 a=candidate:2 2 UDP 1692467198 63.210.102.138 58034 typ srflx raddr 192.168.129.83 rport 63475 m=video 60914 UDP/TLS/RTP/SAVPF 120 c=IN IP4 63.210.102.138 a=rtpmap:120 VP8/90000 a=sendrecv a=candidate:0 1 UDP 2111832319 192.168.129.83 63476 typ host a=candidate:1 1 UDP 1692467199 63.210.102.138 60914 typ srflx raddr 192.168.129.83 rport 63476 a=candidate:2 1 UDP 1692467199 63.210.102.138 63252 typ srflx raddr 192.168.129.83 rport 63476 a=candidate:0 2 UDP 2111832318 192.168.129.83 63477 typ host a=candidate:1 2 UDP 1692467198 63.210.102.138 59956 typ srflx raddr 192.168.129.83 rport 63477 a=candidate:2 2 UDP 1692467198 63.210.102.138 55695 typ srflx raddr 192.168.129.83 rport 63477 SIP/2.0 200 OK Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=54562;received=63.210.102.138;branch=z9hG4bKcxkVDkhksN02zm8U31EiPCdwiTReMKQw From: "Karl"<sip:1000@webrtc.freeswitch.org>;tag=RtwA31FZToUkxXyeZOLy To: <sip:9386@webrtc.freeswitch.org>;tag=tFH27N15gcQyr Contact: <sip:9386@198.22.64.248:5060;transport=udp> Call-ID: eb87cf95-ea84-ae23-d7ee-309431fce563 CSeq: 21413 INVITE Content-Type: application/sdp Content-Length: 1525 User-Agent: FreeSWITCH-mod_sofia/1.5.3b+git~20130708T181922Z~35fca3d290 Accept: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE Supported: precondition,path,replaces Allow-Events: talk,hold,conference,presence,dialog,line-seize,call-info,sla,include-session-description,presence.winfo,message-summary,refer Content-Disposition: session Remote-Party-ID: screen=no;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1373382476 1373382477 IN IP4 198.22.64.248 s=FreeSWITCH c=IN IP4 198.22.64.248 t=0 0 a=msid-semantic: WMS F7UAmqLKFVSItyzNlGnMNjUAopBhjR8O m=audio 24162 RTP/SAVPF 109 101 a=rtpmap:109 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=ptime:20 a=fingerprint:sha-256 06:6C:87:AA:34:0C:96:4E:37:60:F8:9A:42:C2:6B:D3:07:88:ED:59:EF:CA:B5:56:55:2B:3F:C6:3A:0D:06:F6 a=rtcp:24163 IN IP4 198.22.64.248 a=ssrc:975760054 cname:PSe18QBEBkcVJBzU a=ssrc:975760054 msid:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8O a0 a=ssrc:975760054 mslabel:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8O a=ssrc:975760054 label:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8Oa0 a=ice-ufrag:7MwKVhaawChGjfkY a=ice-pwd:XpQW5iqxssi1UHMR a=candidate:6695616536 1 udp 659136 198.22.64.248 24162 typ host generation 0 a=candidate:6695616536 2 udp 659134 198.22.64.248 24163 typ host generation 0 m=video 23786 RTP/SAVPF 120 a=rtpmap:120 VP8/90000 a=fingerprint:sha-256 06:6C:87:AA:34:0C:96:4E:37:60:F8:9A:42:C2:6B:D3:07:88:ED:59:EF:CA:B5:56:55:2B:3F:C6:3A:0D:06:F6 a=rtcp:23787 IN IP4 198.22.64.248 a=ssrc:289067359 cname:PSe18QBEBkcVJBzU a=ssrc:289067359 msid:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8O v0 a=ssrc:289067359 mslabel:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8O a=ssrc:289067359 label:F7UAmqLKFVSItyzNlGnMNjUAopBhjR8Ov0 a=ice-ufrag:VkpGpYNFztvQB068 a=ice-pwd:mf7j9tFZLoAH62BS a=candidate:7819168960 1 udp 659136 198.22.64.248 23786 typ host generation 0 a=candidate:7819168960 2 udp 659134 198.22.64.248 23787 typ host generation 0 *---------------------------------------------------------* -- Thank You: Karl Mack Title: Software Engineer Address: Vail Systems 570 Lake Cook Road Deerfield, IL 60015 Telephone: 1-800-360-8245 Email: kmack(a)vailsys.com

2 4

FreeSWITCH Community Conference Call Today
by Michael Collins 10 Jul '13

10 Jul '13

Hello all, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_10 We are going to be discussing several topics: * ClueCon 2013 <http://www.cluecon.com> - register now and triple your chances to win * New FreeSWITCH wiki site under development * Community discussion of some mailing list questions, like the broker's call<http://lists.freeswitch.org/pipermail/freeswitch-users/2013-July/097491.html>feature Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

1 0

park and blf
by fs＠voice2net.ca 10 Jul '13

10 Jul '13

I have blf's working fine with extensions but I cannot get them to work with parking. Below is the setup I am using to park calls. Any ideas what I need to change to get this working?? Darcy Primrose <extension name="parking_slots"> <condition field="destination_number" expression="^(700\d{1})$" break="on-false"> <action inline="true" application="set" data="slot_count=${fifo(count $1@${domain})}"/> <action inline="true" application="set" data="slot_count=${slot_count:-9:2}"/> </condition> <condition field="${slot_count}" expression="^\:0$" break="always"> <action application="unset" data="fifo_chime_list"/> <action application="set" data="fifo_chime_freq=0"/> <action application="fifo" data="${destination_number}@${domain} in undef local_stream://moh"/> <anti-action application="fifo" data="${destination_number}@${domain} out nowait"/> </condition> </extension>

3 5

Problems with G722 sample rate
by Wesley Akio 10 Jul '13

10 Jul '13

Hi all, I'm having some trouble with G722 and AudioCodes HD310 in latest 1.2-stable... The phone sends the SDP: v=0 o=1000 946688278 946688278 IN IP4 192.168.0.108 s=- c=IN IP4 192.168.0.108 t=0 0 m=audio 4000 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 But Freeswitch is prepared to accept only G722@8000 as can be seen in: [DEBUG] sofia_glue.c:5181 Audio Codec Compare [G722:9:16000:20:64000]/[G722:9:8000:10:64000] For now I've changed the source to accept 16000 but according to Anthony, as per the RFC, 8000 is the correct value. I've seen some conversations between Nandy Dagondon ans Steven Ayre but I'm not sure if it is the same issue, nonetheless I could not find a solution. has anyone faced a similar problem and found a viable solution? Best, Wesley Akio TuntsCorp.com

4 4

Using say in play_and_get_digits
by Vishal Kakkar 10 Jul '13

10 Jul '13

Hi All, I have an IVR like below.. Problem is it repeat complete input recieved from user before asking for confirmation. i.e. user can not just press 1 to proceed while system is repeating recieved inputs. <action application="play_and_get_digits" data="20 21 50 15000 # $${IVR_PROMPTS}/AppVoiceFolder/EnterRechargeDetails.wav $${IVR_PROMPTS}/InvalidInput.wav my_digit ^((\d{10})\*(\d{6})\*(1|3|12))$"/> <action application="playback" data="$${IVR_PROMPTS}/AppVoiceFolder/RepeatEnteredDetails.wav"/> <action application="say" data="hi name_spelled iterated NEUTER ${my_digit}"/> <action application="play_and_get_digits" data="1 1 2 5000 # file_string://$${IVR_PROMPTS}/AppVoiceFolder/Press1ToConfirm.wav!file_string://$${IVR_PROMPTS}/AppVoiceFolder/Press2ToReEnter.wav $${IVR_PROMPTS}/InvalidInput.wav my_choice ^[1-2]"/> Something i tried to achieve this is- <action application="play_and_get_digits" data="20 21 50 15000 # $${IVR_PROMPTS}/AppVoiceFolder/EnterRechargeDetails.wav $${IVR_PROMPTS}/InvalidInput.wav my_digit ^((\d{10})\*(\d{6})\*(1|3|12))$"/> <action application="playback" data="$${IVR_PROMPTS}/AppVoiceFolder/RepeatEnteredDetails.wav"/> <action application="play_and_get_digits" data="1 1 2 5000 # say:'hi name_spelled iterated NEUTER ${my_digit}' file_string://$${IVR_PROMPTS}/AppVoiceFolder/Press1ToConfirm.wav!file_string://$${IVR_PROMPTS}/AppVoiceFolder/Press2ToReEnter.wav $${IVR_PROMPTS}/InvalidInput.wav my_choice ^[1-2]"/> But issue is that it reads "hi name_spelled iterated NEUTER 912345678*10001*12" as string. Please help how can i fix it. Please also note that i have local language sound files kept in "hi" folder so that digits pronounced by system use the same and this is working when i use first method. But when i include say in "play_and_get_digits" it picks up english again. Thanks, -Vishal

1 0