April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Best practice for paging in an environment with multiple-registrations enabled on Sofia profile
by Brian Foster 21 Apr '12

21 Apr '12

As the subject suggests, what is the best way to do paging with multiple-registrations enabled? In my lab I have two phones, one is registered to 2546 and 2686, the other is registered to 2546 and 2738 (so both phones are registered to 2546 and another extension). I have something set up so that FS sends out a page like so: originate {sip_auto_answer=true}user/2546(a)domain.tld&playback(alertmessage.wav) However as you might expect (it does make sense) the first phone to win the lottery and pick up the phone first wins the call and auto answers the call. How do I properly get every extension to playback the message at the same time? -BDF

4 5

How to use alternate presence id for a fifo queue?
by Travis Langhals 21 Apr '12

21 Apr '12

I've setup a fifo park queue based off the example at http://wiki.freeswitch.org/wiki/Park_%26_Retrieve and am trying to use the Aastra BLF/Xfer key to interface with it. The parking fifo queue works great but I have to use two keys, one for BLF to subscribe to queue+<fifo_name> and another for speed dial/tranfer to <fifo_name>. Is there any way to remove the need to have the queue+ prefix needed to subscribe to the fifo presence so that I can just use a single BLF/Xfer key point to <fifo_name>? I found this thread that mentions using the map proto feature to eliminate the need for the prefix but cannot find anything information relating to such a feature. http://jira.freeswitch.org/browse/FS-2831?page=com.atlassian.jira.plugin.sy…

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recommend free/open source sip client for iPhone
by fieldpeak 21 Apr '12

21 Apr '12

Is there any recommended free/open source sip client for iPhone? Thanks. -- Regards, Charles

6 15

Memory usage of mod_local_stream
by Vik Killa 20 Apr '12

20 Apr '12

I'm wondering how many local_stream FS can handle with the following parameters: <param name="channels" value="1"/> <param name="interval" value="20"/> <param name="rate" value="8000"/> <param name="timer-name" value="soft"/> Is there a way to calculate or know how much memory each stream will use with these settings?

1 0

annexb=yes
by Stephen Wilde 20 Apr '12

20 Apr '12

In my Freeswitch, the originated outgoing calls have: Media Attribute (a): fmtp:18 annexb=no How can I change it in annexb=yes ? Stephen

3 4

FS as SBC gets USER_NOT_REGISTERED with registered Users from external
by Karsten Horsmann 20 Apr '12

20 Apr '12

Hi List, I add the register stuff to the public.xml dialplan to force external users to register. i have problems to call registered users. Why i get USER_NOT_REGISTERED? I make an siptrace within Freeswitch and put it here: http://pastebin.freeswitch.org/18893 Thanks. -- Mit freundlichen Grüßen *Karsten Horsmann*

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Re: [Freeswitch-users] SIP 183 on a leg before bleg invite in bridge
by Barnaby Ritchley 20 Apr '12

20 Apr '12

Hi Guys I have a situation where a call is received on a-leg, FS sends 183 session progress to the a-leg before the call is attempted in b-leg. Then, when the b-leg attempts the call for the bridge and sends an invite, it receives 180 ringing, and this is not relayed back through the a leg, so the a-leg never hears ringing. The call goes: Cisco UA (1.1.1.1) -> FS (2.2.2.2) -> Gateway (3.3.3.3) Sofia config has nothing special in it, relevent sections: <param name="inbound-codec-negotiation" value="generous"/> <param name="disable-transcoding" value="true"/> <param name="manual-redirect" value="true"/> <param name="disable-transfer" value="true"/> and the dial plan is straight forward, just a bridge between the two endpoints. Any ideas? Here is the trace of the call: INVITE sip:1234@2.2.2.2:5060 SIP/2.0. Date: Fri, 20 Apr 2012 07:58:36 GMT. Call-Info: <sip:1.1.1.1:5060>;method="NOTIFY;Event=telephone-event;Duration=500". Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY. From: "User" <sip:00000000@1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567. Allow-Events: presence, kpml. Supported: 100rel,timer,resource-priority,replaces. Supported: X-cisco-srtp-fallback. Supported: Geolocation. Min-SE: 1800. Content-Length: 212. To: <sip:1234@2.2.2.2>. Expires: 180. Content-Type: application/sdp. Call-ID: a088a380-f911172c-3d33e6-8d200795(a)1.1.1.1. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979. CSeq: 101 INVITE. Session-Expires: 1800. Max-Forwards: 69. . v=0. o=CiscoSystemsCCM-SIP 2000 1 IN IP4 1.1.1.1. s=SIP Call. c=IN IP4 1.1.1.1. t=0 0. m=audio 31422 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 2012/04/20 08:58:16.142711 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979. From: "User" <sip:00000000@1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567. To: <sip:1234@2.2.2.2>. CSeq: 101 INVITE. User-Agent: FreeSWITCH. Content-Length: 0. . U 2012/04/20 08:58:16.169453 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979. From: "User" <sip:00000000@1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567. To: <sip:1234@2.2.2.2>;tag=vm45yDZjgmDKH. Call-ID: a088a380-f911172c-3d33e6-8d200795(a)1.1.1.1. CSeq: 101 INVITE. Contact: <sip:1234@2.2.2.2:5060;transport=udp>. User-Agent: FreeSWITCH. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 247. Remote-Party-ID: "1234" <sip:1234@2.2.2.2>;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1334881876 1334881877 IN IP4 2.2.2.2. s=FreeSWITCH. c=IN IP4 2.2.2.2. t=0 0. m=audio 26820 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. U 2012/04/20 08:58:16.176126 2.2.2.2:5080 -> 3.3.3.3:5060 INVITE sip:05600660128@3.3.3.3:5060 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5080;rport;branch=z9hG4bK44BS1jH9Za6yS. Max-Forwards: 68. From: "00000000" <sip:00000000@2.2.2.2>;tag=4vjXj69rtpaeS. To: <sip:1234@3.3.3.3:5060>. Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b. CSeq: 27118284 INVITE. Contact: <sip:gw+C4DF4B2A088C91@2.2.2.2:5080;transport=udp;gw=C4DF4B2A088C91>. User-Agent: FreeSWITCH. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 201. X-FS-Support: update_display,send_info. Remote-Party-ID: "00000000" <sip:00000000@2.2.2.2>;party=calling;screen=no;privacy=full. . v=0. o=FreeSWITCH 1334887230 1334887231 IN IP4 2.2.2.2. s=FreeSWITCH. c=IN IP4 2.2.2.2. t=0 0. m=audio 21466 RTP/AVP 8 101 13. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2012/04/20 08:58:16.176573 3.3.3.3:5060 -> 2.2.2.2:5080 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS. From: "00000000" <sip:00000000@2.2.2.2>;tag=4vjXj69rtpaeS. To: <sip:1234@3.3.3.3:5060>. Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b. CSeq: 27118284 INVITE. User-Agent: SBC01. Content-Length: 0. . U 2012/04/20 08:58:16.282935 3.3.3.3:5060 -> 2.2.2.2:5080 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS. From: "00000000" <sip:00000000@2.2.2.2>;tag=4vjXj69rtpaeS. To: <sip:1234@3.3.3.3:5060>;tag=7NNjeSyyjtHSB. Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b. CSeq: 27118284 INVITE. Contact: <sip:05600660128@3.3.3.3:5060;transport=udp>. User-Agent: SBC01. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0.

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Packet-time negotiation problem
by Jose Miguel Sucasas Mejuto 20 Apr '12

20 Apr '12

Hello, I have installed freeSwitch 1.1beta from the git repository (FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000). I have enabled iLBC@30i codec by setting global vars: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM,iLBC@30i,SILK,G729"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,iLBC@30i,SILK,G729"/> I try to call from Bria for iPhone to x-Lite 4.1. Bria offers iLBC@30i: 2012-04-20 00:24:03.740477 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1001(a)sukmad.dyndns-home.com [624007f0-d580-4500-834f-76c2ba578df0] 2012-04-20 00:24:03.740477 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1001(a)sukmad.dyndns-home.com) Running State Change CS_NEW 2012-04-20 00:24:03.740477 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/1001(a)sukmad.dyndns-home.com) State NEW 2012-04-20 00:24:03.740477 [DEBUG] sofia.c:5561 Channel sofia/internal/1001(a)sukmad.dyndns-home.com entering state [received][100] 2012-04-20 00:24:03.740477 [DEBUG] sofia.c:5572 Remote SDP: v=0 o=- 3543863125 3543863125 IN IP4 88.9.239.104 s=cpc_med c=IN IP4 88.9.239.104 t=0 0 m=audio 17002 RTP/AVP 112 18 105 101 c=IN IP4 88.9.239.104 a=rtpmap:112 SILK/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:105 iLBC/8000 a=fmtp:105 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 But FS set the codec to iLBC 98 (according to vars.xml, it corresponds to iLBC@20i): 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[G7221:115:32000:20:48000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[G7221:115:32000:20:48000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[G7221:107:16000:20:32000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[G7221:107:16000:20:32000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[G722:9:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[G722:9:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[PCMU:0:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[PCMU:0:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[PCMA:8:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[PCMA:8:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[GSM:3:8000:20:13200] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[GSM:3:8000:20:13200] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[GSM:3:8000:20:13200] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[iLBC:98:8000:20:15200] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[iLBC:98:8000:20:15200] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[iLBC:98:8000:20:15200] 2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:3006 Set Codec sofia/internal/1001(a)sukmad.dyndns-home.com iLBC/8000 20 ms 160 samples 15200 bits 2012-04-20 00:24:03.740477 [DEBUG] switch_core_codec.c:111 sofia/internal/1001(a)sukmad.dyndns-home.com Original read codec set to iLBC:98 I'm trying to fix the codec in the dialplan by setting absolute_codec_string channel variable: Dialplan: sofia/internal/1001(a)sukmad.dyndns-home.com Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1001(a)sukmad.dyndns-home.com Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1001(a)sukmad.dyndns-home.com Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1001(a)sukmad.dyndns-home.com Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1001(a)sukmad.dyndns-home.com Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/1001(a)sukmad.dyndns-home.com Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1001(a)sukmad.dyndns-home.com Action bridge({absolute_codec_string='iLBC@30i,G729'}user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1001(a)sukmad.dyndns-home.com Action answer() When FS rings in the B-leg, I think it sends an incorrect codec (mode=20) 2012-04-20 00:24:04.170478 [DEBUG] mod_sofia.c:2582 Ring SDP: v=0 o=FreeSWITCH 1334863010 1334863011 IN IP4 88.9.239.104 s=FreeSWITCH c=IN IP4 88.9.239.104 t=0 0 m=audio 11234 RTP/AVP 105 101 a=rtpmap:105 iLBC/8000 a=fmtp:105 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv When I try to answer in B-leg (x-Lite softphone) it offers iLBC 98 (30 ms): 2012-04-20 00:24:13.750480 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/sip:1002@192.168.0.2:3348 [BREAK] 2012-04-20 00:24:13.750480 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/sip:1002@192.168.0.2:3348 [BREAK] 2012-04-20 00:24:13.750480 [DEBUG] sofia.c:5561 Channel sofia/internal/sip:1002@192.168.0.2:3348 entering state [completing][200] 2012-04-20 00:24:13.750480 [DEBUG] sofia.c:5572 Remote SDP: v=0 o=- 1334874336856091 1 IN IP4 192.168.1.2 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.1.2 t=0 0 m=audio 54624 RTP/AVP 98 101 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 and the call ends with the following error: 2012-04-20 00:24:13.800478 [DEBUG] switch_core_session.c:793 Send signal sofia/internal/1001(a)sukmad.dyndns-home.com [BREAK] 2012-04-20 00:24:13.810479 [DEBUG] switch_core_io.c:1095 Engaging Write Buffer at 320 bytes to accommodate 480->320 2012-04-20 00:24:13.980477 [WARNING] mod_sofia.c:1158 Asynchronous PTIME not supported, changing our end from 20 to 30 2012-04-20 00:24:13.980477 [DEBUG] sofia_glue.c:2924 Changing Codec from iLBC@20ms@8000hz to iLBC@30ms@8000hz 2012-04-20 00:24:14.020479 [WARNING] switch_core_codec.c:691 Codec iLBC Exists but not at the desired implementation. 8000hz 30ms 2012-04-20 00:24:14.020479 [ERR] sofia_glue.c:2954 Can't load codec? 2012-04-20 00:24:14.020479 [DEBUG] switch_channel.c:2848 (sofia/internal/1001(a)sukmad.dyndns-home.com) Callstate Change ACTIVE -> HANGUP 2012-04-20 00:24:14.020479 [NOTICE] sofia_glue.c:2955 Hangup sofia/internal/1001(a)sukmad.dyndns-home.com [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Is it possible to re-packetize iLBC streams from iLBC@20i to iLBC@30i?. Why FS choose an incorrect packet time?. I have read about this problem in Internet and I try to uncomment the following line: <param name="rtp-autofix-timing" value="false"/> But when I try to make the call I don't hear nothing (except a big noise)... There are any solution?. I attach all the dump from fs_cli. Thanks in advance. Regards.

1 0

RPMs and Yum repo for EL5 and EL6 based system in beta
by Ken Rice 19 Apr '12

19 Apr '12

Hey guys, Check it out... Yum is on the way for new freeswitch goodness. With root perms (that means logged in as root, su or maybe even sudo) rpm -Uvh http://files.freeswitch.org/yum/freeswitch-release-1-0.noarch.rpm This will install the FreeSwitch YUM repo on your RHEL5 or 6 based box (that includes Centos or Scientific Linux 5 and 6 platforms) Then to install FreeSWITCH with the same configs you get from making samples yum install freeswitch-config-vanilla This will install freeswitch to /usr/local/freeswitch as you are use to doing at this time. RPMs are being rolled every couple of days right now, but will soon be rolled nightly... Find me on #freeswitch on irc.freenode.net as SwK if you have any questions or issues, if you find any bugs please open a Jira and assign it to me. These RPMs are Alpha/Beta quality please test them before using them in a production environment... Additional Packages will be available shortly. Ken aka SwK

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Looking for Kamilio - FreeSWITCH help/consulting
by Abaci 19 Apr '12

19 Apr '12

Hi, We have setup a cluster of multiple FreeSWITCH servers, we are looking to implement Kamailio in the front as a proxy and registration server, I have very little experience with Kamailio, so I'm looking to hire someone with knowledge and experience. please email me abaci64 at gmail dot com Thanks

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