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Re: [Freeswitch-users] mod_sofia@
by Mi Ke 09 Oct '11

09 Oct '11

found it on wiki - it's sip_contact_user variable ----- Original Message ----- From: Mi Ke Sent: 10/08/11 09:33 PM To: Freeswitch Users Subject: [Freeswitch-users] mod_sofia@ Hi ! I originate call from lua script by the following command: local ob_session = freeswitch.Session("{originate_timeout=120,return_ring_ready=true,origination_caller_id_number=" .. calling_id .. ",called_id=" .. called_id .. ",accountcode=" .. accountcode .. ",credit_time=" .. out_credit_time .. ",execute_on_answer=lua b_answer.lua}" .. out_called_id,session) My iface IP is 2.2.2.2, origination_caller_id_number is my calling ID (55555) and out_called_id = sofia/internal/1212121212(a)1.1.1.1 Leg B setup and bridging goes normally, when remote party disconnects, we receive: tport_wakeup_pri(0x81c1a0): events IN tport_recv_event(0x81c1a0) tport_recv_iovec(0x81c1a0) msg 0x7f28386348a0 from (udp/2.2.2.2:5060 http://78.46.44.138:5060/ ) has 425 bytes, veclen = 1 tport_deliver(0x81c1a0): msg 0x7f28386348a0 (425 bytes) from udp/1.1.1.1:5060/sip http://193.104.107.7:5060/sip next=(nil) nta: received BYE sip:mod_sofia@2.2.2.2: http://sip:mod_sofia@78.46.44.138:5060/ 5060 http://sip:mod_sofia@78.46.44.138:5060/ SIP/2.0 (CSeq 48765) nta: canonizing sip:mod_sofia@2.2.2.2:5060 http://sip:mod_sofia@78.46.44.138:5060/ with contact nta: BYE (48765) going to existing leg nua: nua_stack_process_request: entering nua(0x94c4a0): event i_bye 100 Trying nua: nua_application_event: entering 2011-10-08 14:54:48.866309 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/ 841266482056(a)193.104.107.7 .1.1.1 [BREAK] nua: nua_handle_magic: entering 2011-10-08 14:54:48.886307 [DEBUG] switch_channel.c:2800 (sofia/internal/ 841266482056(a)193.104.107.7 .1.1.1) Callstate Change ACTIVE -> HANGUP (normal bye handshake follows...) Should remote party see me as sip:55555@2.2.2.2 and not as sip:mod_sofia@2.2.2.2 ? What info should I include into session originate to acheive that? Thanks / Mike

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Failed to get callee number during the call
by fieldpeak 09 Oct '11

09 Oct '11

Hello Friends, i would like to get the callee number during the call, i used the function 'switch_channel_get_variable', however, when i display it to the console, it shows 'null', i tried call this func before and after answering the call, both failed, could anyone help advise if i used the correct function or how to realzize it, thanks. source code as below, const char* in_calledstation; in_calledstation = switch_channel_get_variable(channel, "sip_callee_id_number"); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "in_username: %s, in_calledstation: %s, in_uniqueid: %s, sessiontime: %d \n", in_username, in_calledstation, in_uniqueid, sessiontime); -- Regards, Charles

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Host registration on shared ODBC
by Madovsky 09 Oct '11

09 Oct '11

I got a strange behaviour recently. if a phone is registrered on node 1 and check every 20s with "sofia contact" so this command returns the right node. but if run "sofia status profile myprofile" it returns the wrong node after an incoming call. there are nothing abnormal on logs at levevl7. I run FS with postgresql and shared ODBC. FreeSWITCH Version 1.0.head (git-f4fa0c0 2011-10-04 20-17-12 -0500) Is anyone had already this problem ? Thanks Franck

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Newbie question...
by Francisco Javier Cintrón Olguín 09 Oct '11

09 Oct '11

Ok guys, I am a newbie in VOIP stuff and in freeswitch of course. I got the freeswitch book and installed freeswitch 1.0.6 in my windows 7 pro rig. I also installed XLite 4 but when I try to register the extension is telling me: "Acount failed to enable" SIP 408 error. I am sure I stoped my firewall and even with that I am getting same error. I have not changed anything from the default configuration, the extension I am trying to register is extension 1000, with password 1234. ¿Could you help me please? Thanks in advance.

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Short Calls - how to prevent?
by Avi Marcus 09 Oct '11

09 Oct '11

Looking at my calling card logs for the last month, I have nearly 20% of my calls under 6 seconds. I know a LOT of them are from hanging up after 20+ seconds, presumably because they got voicemail. Most carriers don't like this and have heavy fees - e.g. 2cents for short calls, or just 2c for each short call over the limit. Hmm, for my call volume that's only a few dollars... I wonder if anyone had any ideas? I was thinking a transfer_on_hangup for leg B to send to a lua script - if the duration is under 6 seconds, then sleep for a second a check again. But I suppose only charge the caller for the first 6 seconds (since I never said 12 second minimum). (like 30cents instead of $5 fee on 900 short calls, of 5k calls). Actually, 240 of them might be because of the ANI I was passing... Does anyone have other ideas, or already wrote this code? Thanks, -Avi Marcus 718-989-9485 (USA) 054-844-3271 (Israel Kosher) 077-228-5055 (Israel Landline) 020-3519-3606 (UK)

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Bridge calls to Java ESL socket does not work as expected
by xl127 08 Oct '11

08 Oct '11

Hello, I am trying to bridge multiple calls to multiple Java ESL outbound socket applications. Basically I want to check if the first socket app is engaged then the call should be transferred to the second socket app and so on. For initial test, I tried two concurrent calls, the problem is both calls are always answered by the first outbound socket. I have following setting in the dialplan: <extension name="sipgateDialedIn"> <condition field="destination_number" expression="^5800$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="loopback/5801"/> <action application="bridge" data="loopback/5802"/> <action application="set" data="tts_engine=flite"/> <action application="set" data="tts_voice=kal"/> <action application="speak" data="Sorry, I am busy at the moment. Please try again later."/> <action application="sleep" data="10"/> <action application="hangup"/> </condition> </extension> <extension name="myOutboundESL5801"> <condition field="destination_number" expression="^5801$"> <action application="socket" data="127.0.0.1:9981 async full"/> </condition> </extension> <extension name="myOutboundESL5802"> <condition field="destination_number" expression="^5802$"> <action application="socket" data="127.0.0.1:9982 async full"/> </condition> </extension> I expect the first call to 5800 would be asnwered by extension 5801, the second concurrent call to 5800 will be answered by extension 5802, and the third call gets a busy message. I use Java ESL client-0.9.2 libraries. How do I check in the dialplan if the socket is engaged? I've tried and googled a while but without success. Any advices please? Many thanks! Xing -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012

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mod_sofia@
by Mi Ke 08 Oct '11

08 Oct '11

Hi ! I originate call from lua script by the following command: local ob_session = freeswitch.Session("{originate_timeout=120,return_ring_ready=true,origination_caller_id_number=" .. calling_id .. ",called_id=" .. called_id .. ",accountcode=" .. accountcode .. ",credit_time=" .. out_credit_time .. ",execute_on_answer=lua b_answer.lua}" .. out_called_id,session) My iface IP is 2.2.2.2, origination_caller_id_number is my calling ID (55555) and out_called_id = sofia/internal/1212121212(a)1.1.1.1 Leg B setup and bridging goes normally, when remote party disconnects, we receive: tport_wakeup_pri(0x81c1a0): events IN tport_recv_event(0x81c1a0) tport_recv_iovec(0x81c1a0) msg 0x7f28386348a0 from (udp/2.2.2.2:5060 http://78.46.44.138:5060/ ) has 425 bytes, veclen = 1 tport_deliver(0x81c1a0): msg 0x7f28386348a0 (425 bytes) from udp/1.1.1.1:5060/sip http://193.104.107.7:5060/sip next=(nil) nta: received BYE sip:mod_sofia@2.2.2.2: http://sip:mod_sofia@78.46.44.138:5060/ http://sip:mod_sofia@78.46.44.138:5060/ 5060 http://sip:mod_sofia@78.46.44.138:5060/ SIP/2.0 (CSeq 48765) nta: canonizing sip:mod_sofia@2.2.2.2:5060 http://sip:mod_sofia@78.46.44.138:5060/ with contact nta: BYE (48765) going to existing leg nua: nua_stack_process_request: entering nua(0x94c4a0): event i_bye 100 Trying nua: nua_application_event: entering 2011-10-08 14:54:48.866309 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/ 841266482056(a)193.104.107.7 .1.1.1 [BREAK] nua: nua_handle_magic: entering 2011-10-08 14:54:48.886307 [DEBUG] switch_channel.c:2800 (sofia/internal/ 841266482056(a)193.104.107.7 .1.1.1) Callstate Change ACTIVE -> HANGUP (normal bye handshake follows...) Should remote party see me as sip:55555@2.2.2.2 and not as sip:mod_sofia@2.2.2.2 ? What info should I include into session originate to acheive that? Thanks / Mike

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Re: [Freeswitch-users] Why does group_confirm_file 3 times from?
by Thomas Hoellriegel 08 Oct '11

08 Oct '11

Hi guys, I'm sorry, because my ISP was temporarily unavailable, have been lost some mails to me. I have installed the git version from fs. But the Problem is the same. I connect via sip. My Extension is: <extension name="callscreen"> <condition field="destination_number" expression="^callscreen"> <action application="set" data="group_confirm_file=playback $${base_dir}/sounds/anrufe/achtung-dringend-call.wav"/> <action application="set" data="group_confirm_key=exec"/> <action application="bridge" data="loopback/0/operator1"/> </condition> </extension> Fs plays before connecting this file 3 times from. No matter what digit I press. Fs repeate the soundfile 3 times. I don.t find a parameter to turn off this. Can someone help me please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

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Desktop phone - answer a ringing call via API call
by Weigel, Stefan 08 Oct '11

08 Oct '11

Hi all, Is there a way to answer a ringing desktop phone (Polycom Soundpoint IP 560) via api command ? I tried with SendMsg <uuid> and 'answer' application, but it's not want I need/want. Thanks and best regards, Stefan

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Call bridge problem
by Chad Vogel 07 Oct '11

07 Oct '11

Hello, I'm having a problem bridging out calls, FS is sending 500 Internal Server Error to our gateway. Any thoughts why this is? <include> <context name="public"> <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> <extension name="outside_call" continue="true"> <condition> <action application="set" data="outside_call=true"/> <action application="set" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/> </condition> </extension> <extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> </condition> </extension> <extension name="4142211800"> <condition field="destination_number" expression="^(\+?14142211800)$"> <action application="set" data="effective_caller_id_name=LyonL"/> <action application="set" data="effective_caller_id_number=+14142211800"/> <action application="bridge" data="sofia/gateway/L3CN/+15618911806"/> </condition> </extension> </context> </include> <profile name="external">   <gateways> <gateway name="L3CN"> <param name="username" value="1-F2la9"/> <param name="password" value="password"/> <param name="realm" value="BroadWorks"/> <param name="proxy" value="4.55.35.60:5070"/> <param name="from-domain" value="69.65.114.181:5060"/> <param name="dtmf-type" value="rfc2833"/> <param name="extension-in-contact" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="register" value="false"/> </gateway> </gateways> <aliases>  </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/>   <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/>    <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <param name="user-agent-string" value="LyonL"/>          <param name="inbound-codec-negotiation" value="greedy"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="NDLB-force-rport" value="server-only"/>  <param name="rtp-ip" value="$${external_rtp_ip}"/> <param name="sip-ip" value="$${external_sip_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/>   <param name="tls" value="$${external_ssl_enable}"/>  <param name="tls-bind-params" value="transport=tls"/>  <param name="tls-sip-port" value="$${external_tls_port}"/>  <param name="tls-cert-dir" value="$${external_ssl_dir}"/>  <param name="tls-version" value="$${sip_tls_version}"/> </settings> </profile> send 1283 bytes to udp/[4.55.35.60]:5070 at 19:50:28.998153: ------------------------------------------------------------------------ INVITE sip:+15618911806@4.55.35.60:5070 SIP/2.0 Via: SIP/2.0/UDP 69.65.114.181;rport;branch=z9hG4bK62S7Fr2Um459j Max-Forwards: 8 From: "LyonL" <sip:+14142211800@69.65.114.181:5060>;tag=K0gUSK5HNr3US To: <sip:+15618911806@4.55.35.60:5070> Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276 CSeq: 18629251 INVITE Contact: <sip:1-F2la9@69.65.114.181:5060;transport=udp;gw=L3CN> Expires: 3600 User-Agent: LyonL Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE FER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Authorization: Digest username="1-F2la9", realm="BroadWorks", nonce="BroadWor ksXgtg5vbouT4lpitdBW", cnonce="SKxoW2r3Ei+9o+X8ky5idg", algorithm=MD5, uri="sip: +15618911806@4.55.35.60:5070", response="9f0325808936e22ae050921c176f2730", qop= auth, nc=00000001 Content-Type: application/sdp Content-Disposition: session Content-Length: 229 X-FS-Support: update_display Remote-Party-ID: "LyonL" <sip:+14142211800@69.65.114.181:5060>;party=calling; screen=yes;privacy=off v=0 o=FreeSWITCH 1317910968 1317910969 IN IP4 69.65.114.181 s=FreeSWITCH c=IN IP4 69.65.114.181 t=0 0 m=audio 19660 RTP/AVP 0 8 18 101 13 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 297 bytes from udp/[4.55.35.60]:5070 at 19:50:29.002153: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 69.65.114.181;received=69.65.114.181;branch=z9hG4bK62S7Fr2Um 459j;rport=5060 From: "LyonL" <sip:+14142211800@69.65.114.181:5060>;tag=K0gUSK5HNr3US To: <sip:+15618911806@4.55.35.60:5070> Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276 CSeq: 18629251 INVITE ------------------------------------------------------------------------ recv 468 bytes from udp/[4.55.35.60]:5070 at 19:50:29.008153: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 69.65.114.181;received=69.65.114.181;branch=z9hG4bK62S7Fr2Um 459j;rport=5060 From: "LyonL" <sip:+14142211800@69.65.114.181:5060>;tag=K0gUSK5HNr3US To: <sip:+15618911806@4.55.35.60:5070>;tag=SDcpmkf99-1282242359-1317930631679 Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276 CSeq: 18629251 INVITE WWW-Authenticate: DIGEST qop="auth",nonce="BroadWorksXgtg5vbpbTeh3n5wBW",algo rithm=MD5,realm="BroadWorks" Content-Length: 0 ------------------------------------------------------------------------ send 367 bytes to udp/[4.55.35.60]:5070 at 19:50:29.008153: ------------------------------------------------------------------------ ACK sip:+15618911806@4.55.35.60:5070 SIP/2.0 Via: SIP/2.0/UDP 69.65.114.181;rport;branch=z9hG4bK62S7Fr2Um459j Max-Forwards: 8 From: "LyonL" <sip:+14142211800@69.65.114.181:5060>;tag=K0gUSK5HNr3US To: <sip:+15618911806@4.55.35.60:5070>;tag=SDcpmkf99-1282242359-1317930631679 Call-ID: 48a9a931-6af7-122f-a3bd-e5fc932e6276 CSeq: 18629251 ACK Content-Length: 0 ------------------------------------------------------------------------ 2011-10-06 19:50:28.979152 [NOTICE] sofia.c:5897 Hangup sofia/external/+15618911 806 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED] 2011-10-06 19:50:28.979152 [INFO] mod_dptools.c:2810 Originate Failed. Cause: N ORMAL_UNSPECIFIED 2011-10-06 19:50:28.979152 [NOTICE] mod_dptools.c:2929 Hangup sofia/external/+15 618911806(a)4.55.35.60<mailto:618911806@4.55.35.60> [CS_EXECUTE] [NORMAL_UNSPECIFIED] send 380 bytes to udp/[4.55.35.60]:5070 at 19:50:29.010153: ------------------------------------------------------------------------ SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 4.55.35.60:5070;branch=z9hG4bKtil5pp004o3hvjs9n2c1.1 From: <sip:+15618911806@4.55.35.60;user=phone>;tag=SDesfje01-1691283656-13179 30631601- To: "4142211800 4142211800" <sip:+14142211800@69.65.114.181>;tag=jQQ2QrmerFD9 D Call-ID: SDesfje01-bc0436bf1c3c496f2c13c86f66280e15-v3000i1 CSeq: 761577689 INVITE Content-Length: 0 ------------------------------------------------------------------------ 2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1367 Session 19 (sofia /external/+15618911806(a)4.55.35.60) Ended 2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1369 Close Channel sof ia/external/+15618911806(a)4.55.35.60<mailto:ia/external/+15618911806@4.55.35.60> [CS_DESTROY] 2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1367 Session 20 (sofia /external/+15618911806) Ended 2011-10-06 19:50:28.979152 [NOTICE] switch_core_session.c:1369 Close Channel sof ia/external/+15618911806 [CS_DESTROY]

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