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Mod_SPANDSP
by Anto 23 Jan '12

23 Jan '12

hello I'm adapting the files are in http://wiki.freeswitch.org/wiki/Mod_spandsp to make faxing my business. I wanted to know whether to send the fax, an error occurs (destination busy, etc.), this is still retrying successive times or is there a way to handle this. I've been looking fax_result_code and fax_result_text, to make a correct treatment but have not found information on them. Does anyone have any examples or can guide me? Thank you very much Best Regards Anto

2 2

Freeswitch + Freetdm errors
by Carlos Cesario 23 Jan '12

23 Jan '12

Hello guys, In randon moments when I start the freeswitch service, I get the following errors (patebin link)... This is as infinit loop. And I need restart the service 5 times to it UP. In my try to discovery the problem, I believe that the problem happen when my E1 link cable is disconnected. http://pastebin.freeswitch.org/18203 Somebody have idea about this ? Greats Carlos

1 0

Response Custom header in CDR
by Carlo Dimaggio 23 Jan '12

23 Jan '12

Hi all, I need to store a custom response header in CDR. Let me explain the scenario. I have a Kamailio (SIP Proxy) that sends invite toFreeswitch (B2BUA)for call processing (prepaid/CDR); the call is then relayed back to kamailio for LCR (it choose the right destination/carrier): user -> kamailio -> freeswitch -> kamailio -> provider I can set a custom header in the reply route of kamailio (X-Provider: 1.2.3.4) but I don't know how take this information in freeswitch... Could ${sip_ph_X-Provider}/${sip_rh_X-Provider}/${sip_bye_X-Provider} CDR? user <- kamailio <- freeswitch <- kamailio (X-Provider) <- provider (100 or 200 response code) I want to store this information in order to do statistics and profits tracking. Best Regards,

1 0

Problems with setinputCallback
by Dinesh Kapoor 23 Jan '12

23 Jan '12

Hello, I have a functionality that I am trying to implement via Freeswitch and mod_python, in which a user calls in, Freeswitch hangs up the call and then call the user back. All should happen from inside Freeswitch. Here is the code that I had written for it: def handler(session, args): if is_callback_set(app_instance): schedule_callback(app_instance, session.getCallerID()) is_callback_set(app_instance) : checks if we need to callback on this application instance. schedule_callback(app_instance, session.getCallerID()) : function to call the caller back inside schedule_callback, I create a new thread called CallerbackThread, and start the callback in another thread. Inside CallBackThread, following happens: def run(self): new_session = Session(url) if new_session.answered(): new_session.ai = self.app_instance new_session.getCallerID = lambda: new_session.getVariable("destination_number") new_session.setInputCallback(call_back) new_session.isHungup = lambda: new_session.getState() == 'CS_HANGUP' while not new_session.isHungup(): new_session.streamFile("abcd.mp3") new_session.hangup() def call_back(session,what,obj): logger.debug("hi callback") The originate occurs and I am able to hear the file getting streamed correctly. But when I press any dtmf input, then on first input nothing happens, and in the next input I start getting "TypeError : call_back expects 3 args, 0 given" error. I dont know what can be the solution. Any help regarding this is highly appreciated! Regards, Dinesh

1 0

Trigger re-INVITE with different SDP during a call
by Roland Hänel 23 Jan '12

23 Jan '12

Hi, Is there a way to trigger a SIP re-INVITE for a channel that has already been answered, and to make FreeSwitch offer a certain list of codecs to the endpoint? Let's assume the call is answered with PCMA/PCMU, then the user enters an IVR, now we know that he's going to join a conference. For the conference, I might want to enable G.722 and H.264 video (which should not generally be enabled for all incoming calls). I found stuff about codec-related channel variables, but no 'trigger' to tell FS to make a codec re-negotiation. Greetings, Roland

4 4

Profiles and gateways
by David Villasmil 23 Jan '12

23 Jan '12

Hello All, I'm receiving calls on, say IP 1.2.3.4 and i want to send them out on IP 1.2.3.4 or IP 5.6.7.8 depending on the gateway I want to use. For this, i created TWO profiles, one binding to IP1, the other to IP2. Then I created one gateway on each profile. BUT, FS is loading both GWs on BOTH PROFILES! Is this normal? Can anyone give me a hand with this? Thanks! David my fs-cli output: freeswitch@127.0.0.1@internal> sofia xmlstatus <?xml version="1.0" encoding="ISO-8859-1"?> <profiles> <profile> <name>IP_REAL</name> <type>profile</type> <data>sip:mod_sofia@1.2.3.4:5060</data> <state>RUNNING (0)</state> </profile> <gateway> <name>example.com</name> <type>gateway</type> <data>sip:@example.com</data> <state>NOREG</state> </gateway> <gateway> <name>test</name> <type>gateway</type> <data>sip:@Y.Y.Y.Y</data> <state>NOREG</state> </gateway> <gateway> <name>GREAT</name> <type>gateway</type> <data>sip:@X.X.X.X</data> <state>NOREG</state> </gateway> <profile> <name>IP_SPAIN_1</name> <type>profile</type> <data>sip:mod_sofia@5.6.7.8:5060</data> <state>RUNNING (0)</state> </profile> <gateway> <name>example.com</name> <type>gateway</type> <data>sip:@example.com</data> <state>NOREG</state> </gateway> <gateway> <name>test</name> <type>gateway</type> <data>sip:@Y.Y.Y.Y</data> <state>NOREG</state> </gateway> <gateway> <name>GREAT</name> <type>gateway</type> <data>sip:@X.X.X.X</data> <state>NOREG</state> </gateway> </profiles> My sofia: <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/>  <param name="debug-presence" value="0"/> </global_settings> <profiles> <profile name="IP_REAL">   <gateways> <gateway name="test"> <param name="realm" value="Y.Y.Y.Y"/> <param name="username" value=""/> <param name="password" value=""/> <param name="register" value="false"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </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="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/>  <param name="dialplan" value="XML"/> <param name="context" value="IP_REAL"/> <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="$${hold_music}"/> <param name="rtp-timer-name" value="soft"/>   <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/>         <param name="apply-nat-acl" value="rfc1918"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/>  <param name="rtp-ip" value="1.2.3.4"/> <param name="sip-ip" value="1.2.3.4"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="3600"/> <param name="rtp-hold-timeout-sec" value="1800"/>   <param name="tls" value="$${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="$${sip_tls_version}"/> </settings> </profile> <profile name="IP_SPAIN_1">   <gateways> <gateway name="GREAT"> <param name="realm" value="X.X.X.X"/> <param name="username" value=""/> <param name="password" value=""/> <param name="register" value="false"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </gateways> <aliases>  </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="debug" value="0"/>   <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/>  <param name="dialplan" value="XML"/> <param name="context" value="IP_SPAIN_1"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${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="apply-nat-acl" value="rfc1918"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/>  <param name="rtp-ip" value="5.6.7.8"/> <param name="sip-ip" value="5.6.7.8"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="3600"/> <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> </profiles> </configuration>

2 2

VoIP SIP SDK with Cisco
by thomasirvin 23 Jan '12

23 Jan '12

Hi everyone, I have a small company and would like to use VoIP service with my Cisco System. Currently I am testing the SDK of Ozeki and it works fine with Cisco Unified CM (http://www.voip-sip-sdk.com/p_42-how-to-setup-ozeki-voip-sip-sdk-with-cisco…) so far. Has anyone experienced anything with this solution after a longer test period? Thanks in advance. Thomas -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/VoIP-SIP-SDK-with-Cisco-tp721… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

G.722 negotiation issue with AVM Fritzbox
by Roland Hänel 23 Jan '12

23 Jan '12

Hi, It seems that FreeSwitch cannot negotiate G.722 with an AVM Fritzbox device (*the* most popular VoIP box/router in Germany) because of what seems to be a concatenation of several bugs. First of all, the FritzBox sends this SDP in the INVITE: INVITE sip:+4921173062190950@192.168.232.164:6060;transport=udp;gw=duro SIP/2.0. [...] v=0. o=HuaweiSoftX3000 8451448 8451448 IN IP4 213.148.136.222. s=Sip Call. c=IN IP4 213.148.136.222. t=0 0. m=audio 28800 RTP/AVP 9 8 0 102 100 99 97. *a=rtpmap:9 G722/16000.* a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:102 G726-32/8000. a=rtpmap:100 G726-40/8000. a=rtpmap:99 G726-24/8000. a=rtpmap:97 iLBC/8000. a=ptime:30. a=fmtp:97 mode=30. which is wrong since RFC 3551, Section 4.5.2 clearly states that the RTP clock is 8kHz. We could all blame it on AVM here, but when taking a look at the "200 OK" from FreeSwitch, that's really not what should happen: SIP/2.0 200 OK. [...] v=0. o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164. s=FreeSWITCH. c=IN IP4 192.168.232.164. t=0 0. m=audio *0* RTP/AVP 96. *a=rtpmap:96 G722/8000.* So FreeSwitch announces G.722/8000 (correct), but as RTP/AVP ID 96 (9 was requested) and with *UDP Port 0*. The FritzBox BYE's the call when it receives that. What's even more funny is that in the FreeSwitch debugging, it says about the local SDP: [...] 2012-01-20 17:01:04.344716 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/external/+4921653258007(a)qsc.de] 192.168.232.164 port 23418 -> 213.148.136.222 port 28800 codec: 9 ms: 20 2012-01-20 17:01:04.344716 [DEBUG] switch_rtp.c:1659 Starting timer [soft] 160 bytes per 20ms 2012-01-20 17:01:04.344716 [DEBUG] mod_sofia.c:750 Local SDP sofia/external/+4921653258007(a)qsc.de: v=0 o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164 s=FreeSWITCH c=IN IP4 192.168.232.164 t=0 0 *m=audio 23418 RTP/AVP 9* *a=rtpmap:9 G722/8000* a=silenceSupp:off - - - - a=ptime:20 a=sendrecv So there, everything is correct. Does anyone have an idea about this one? Actually, I have taken a look at the source but I was not able to find the point where the UDP port gets lost... The full debugging trace and packet capture ist attached. Greetings, Roland # U 2012/01/20 17:01:04.323112 213.148.136.222:5060 -> 192.168.232.164:6060 INVITE sip:+4921173062190950@192.168.232.164:6060;transport=udp;gw=duro SIP/2.0. Via: SIP/2.0/UDP 213.148.136.222:5060;branch=z9hG4bK7u3e5920b061csoit5d0.1. Call-ID: SD24vi901-73c7d5e35e297c6dd1cbccbf5ab175ee-l65h8l3. To: <sip:+4921173062190950@qsc.de>. From: "+4921653258007" <sip:+4921653258007@qsc.de >;tag=SD24vi901-88lt6slg-CC-35. CSeq: 1 INVITE. Max-Forwards: 65. P-Asserted-Identity: <sip:+4921653258007@qsc.de>. Contact: <sip:021653258007@213.148.136.222:5060;transport=udp>. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. User-Agent: Huawei SoftX3000 V300R010. Supported: 100rel. Content-Length: 352. Content-Type: application/sdp. P-Called-Party-ID: sip:+4921173062190950@qsc.de. X-ORIGINAL-DDI-URI: sip:+4921173062190950@qsc.de. X-CID: mrg5ml6g71ggh5tty7yh7hc5osy5os8y@SoftX3000. . v=0. o=HuaweiSoftX3000 8451448 8451448 IN IP4 213.148.136.222. s=Sip Call. c=IN IP4 213.148.136.222. t=0 0. m=audio 28800 RTP/AVP 9 8 0 102 100 99 97. a=rtpmap:9 G722/16000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:102 G726-32/8000. a=rtpmap:100 G726-40/8000. a=rtpmap:99 G726-24/8000. a=rtpmap:97 iLBC/8000. a=ptime:30. a=fmtp:97 mode=30. # U 2012/01/20 17:01:04.323965 192.168.232.164:6060 -> 213.148.136.222:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 213.148.136.222:5060;branch=z9hG4bK7u3e5920b061csoit5d0.1. From: "+4921653258007" <sip:+4921653258007@qsc.de >;tag=SD24vi901-88lt6slg-CC-35. To: <sip:+4921173062190950@qsc.de>. Call-ID: SD24vi901-73c7d5e35e297c6dd1cbccbf5ab175ee-l65h8l3. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-cfa926d 2012-01-10 17-33-40 -0600. Content-Length: 0. . # U 2012/01/20 17:01:04.354937 192.168.232.164:6060 -> 213.148.136.222:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 213.148.136.222:5060;branch=z9hG4bK7u3e5920b061csoit5d0.1. From: "+4921653258007" <sip:+4921653258007@qsc.de >;tag=SD24vi901-88lt6slg-CC-35. To: <sip:+4921173062190950@qsc.de>;tag=ejS9yr5v3a6Qj. Call-ID: SD24vi901-73c7d5e35e297c6dd1cbccbf5ab175ee-l65h8l3. CSeq: 1 INVITE. Contact: <sip:+4921173062190950@192.168.232.164:6060;transport=udp>. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-cfa926d 2012-01-10 17-33-40 -0600. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 156. P-Asserted-Identity: "+4921173062190950" <sip:+4921173062190950@qsc.de>. . v=0. o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164. s=FreeSWITCH. c=IN IP4 192.168.232.164. t=0 0. m=audio 0 RTP/AVP 96. a=rtpmap:96 G722/8000. ------------------------ 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_NEW 2012-01-20 17:01:04.324734 [DEBUG] sofia.c:5494 Channel sofia/external/+ 4921653258007(a)qsc.de entering state [received][100] 2012-01-20 17:01:04.324734 [DEBUG] sofia.c:5505 Remote SDP: v=0 o=HuaweiSoftX3000 8451448 8451448 IN IP4 213.148.136.222 s=Sip Call c=IN IP4 213.148.136.222 t=0 0 m=audio 28800 RTP/AVP 9 8 0 102 100 99 97 a=rtpmap:9 G722/16000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:100 G726-40/8000 a=rtpmap:99 G726-24/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=ptime:30 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:380 (sofia/external/+4921653258007(a)qsc.de) State NEW 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G722:9:16000:30:64000]/[G722:9:8000:20:64000] 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:4814 Bah HUMBUG! Sticking with G722@8000h@20i 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:2919 Set Codec sofia/external/+4921653258007(a)qsc.de G722/8000 20 ms 160 samples 64000 bits 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:4930 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2012-01-20 17:01:04.324734 [DEBUG] sofia.c:5717 (sofia/external/+ 4921653258007(a)qsc.de) State Change CS_NEW -> CS_INIT 2012-01-20 17:01:04.324734 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_INIT 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:401 (sofia/external/+4921653258007(a)qsc.de) State INIT 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:85 sofia/external/+ 4921653258007(a)qsc.de SOFIA INIT 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:125 (sofia/external/+ 4921653258007(a)qsc.de) State Change CS_INIT -> CS_ROUTING 2012-01-20 17:01:04.324734 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:401 (sofia/external/+4921653258007(a)qsc.de) State INIT going to sleep 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_ROUTING 2012-01-20 17:01:04.324734 [DEBUG] switch_channel.c:1890 (sofia/external/+ 4921653258007(a)qsc.de) Callstate Change DOWN -> RINGING 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:410 (sofia/external/+4921653258007(a)qsc.de) State ROUTING 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:148 sofia/external/+ 4921653258007(a)qsc.de SOFIA ROUTING 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:104 sofia/external/+4921653258007(a)qsc.de Standard ROUTING 2012-01-20 17:01:04.324734 [INFO] mod_dialplan_xml.c:481 Processing +4921653258007 <+4921653258007>->+4921173062190950 in context default Dialplan: sofia/external/+4921653258007(a)qsc.de parsing [default->default] continue=false Dialplan: sofia/external/+4921653258007(a)qsc.de Absolute Condition [default] Dialplan: sofia/external/+4921653258007(a)qsc.de Action park() 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:154 (sofia/external/+4921653258007(a)qsc.de) State Change CS_ROUTING -> CS_EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:410 (sofia/external/+4921653258007(a)qsc.de) State ROUTING going to sleep 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:417 (sofia/external/+4921653258007(a)qsc.de) State EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:241 sofia/external/+ 4921653258007(a)qsc.de SOFIA EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:192 sofia/external/+4921653258007(a)qsc.de Standard EXECUTE EXECUTE sofia/external/+4921653258007(a)qsc.de park() 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.344716 [DEBUG] switch_ivr.c:591 sofia/external/+ 4921653258007(a)qsc.de Command Execute answer() EXECUTE sofia/external/+4921653258007(a)qsc.de answer() 2012-01-20 17:01:04.344716 [INFO] switch_nat.c:590 NAT port mapping disabled 2012-01-20 17:01:04.344716 [INFO] switch_nat.c:590 NAT port mapping disabled 2012-01-20 17:01:04.344716 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/external/+4921653258007(a)qsc.de] 192.168.232.164 port 23418 -> 213.148.136.222 port 28800 codec: 9 ms: 20 2012-01-20 17:01:04.344716 [DEBUG] switch_rtp.c:1659 Starting timer [soft] 160 bytes per 20ms 2012-01-20 17:01:04.344716 [DEBUG] mod_sofia.c:750 Local SDP sofia/external/+4921653258007(a)qsc.de: v=0 o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164 s=FreeSWITCH c=IN IP4 192.168.232.164 t=0 0 m=audio 23418 RTP/AVP 9 a=rtpmap:9 G722/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:729 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.344716 [DEBUG] switch_channel.c:3194 (sofia/external/+ 4921653258007(a)qsc.de) Callstate Change RINGING -> ACTIVE 2012-01-20 17:01:04.344716 [NOTICE] mod_dptools.c:1135 Channel [sofia/external/+4921653258007(a)qsc.de] has been answered 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.344716 [DEBUG] sofia.c:5494 Channel sofia/external/+ 4921653258007(a)qsc.de entering state [completed][200] 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.364714 [DEBUG] switch_ivr.c:591 sofia/external/+ 4921653258007(a)qsc.de Command Execute playback(silence_stream://100) EXECUTE sofia/external/+4921653258007(a)qsc.de playback(silence_stream://100) 2012-01-20 17:01:04.364714 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16@16000hz 1 channels 20ms 2012-01-20 17:01:04.464734 [DEBUG] switch_ivr_play_say.c:1678 done playing file silence_stream://100 2012-01-20 17:01:04.464734 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.484715 [DEBUG] switch_ivr.c:591 sofia/external/+ 4921653258007(a)qsc.de Command Execute sleep(900) EXECUTE sofia/external/+4921653258007(a)qsc.de sleep(900) 2012-01-20 17:01:04.624743 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.624743 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.624743 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.644728 [DEBUG] sofia.c:5494 Channel sofia/external/+ 4921653258007(a)qsc.de entering state [ready][200] 2012-01-20 17:01:04.664735 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_channel.c:2852 (sofia/external/+ 4921653258007(a)qsc.de) Callstate Change ACTIVE -> HANGUP 2012-01-20 17:01:04.684730 [NOTICE] sofia.c:623 Hangup sofia/external/+ 4921653258007(a)qsc.de [CS_EXECUTE] [NORMAL_CLEARING] 2012-01-20 17:01:04.684730 [DEBUG] switch_channel.c:2875 Send signal sofia/external/+4921653258007(a)qsc.de [KILL] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:2285 sofia/external/+4921653258007(a)qsc.de skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:2285 sofia/external/+4921653258007(a)qsc.de skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:417 (sofia/external/+4921653258007(a)qsc.de) State EXECUTE going to sleep 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_HANGUP 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:602 (sofia/external/+4921653258007(a)qsc.de) State HANGUP 2012-01-20 17:01:04.684730 [DEBUG] mod_sofia.c:463 sofia/external/+ 4921653258007(a)qsc.de Overriding SIP cause 480 with 200 from the other leg 2012-01-20 17:01:04.684730 [DEBUG] mod_sofia.c:469 Channel sofia/external/+ 4921653258007(a)qsc.de hanging up, cause: NORMAL_CLEARING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:47 sofia/external/+4921653258007(a)qsc.de Standard HANGUP, cause: NORMAL_CLEARING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:602 (sofia/external/+4921653258007(a)qsc.de) State HANGUP going to sleep 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:393 (sofia/external/+4921653258007(a)qsc.de) State Change CS_HANGUP -> CS_REPORTING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_REPORTING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:662 (sofia/external/+4921653258007(a)qsc.de) State REPORTING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:79 sofia/external/+4921653258007(a)qsc.de Standard REPORTING, cause: NORMAL_CLEARING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:662 (sofia/external/+4921653258007(a)qsc.de) State REPORTING going to sleep 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:387 (sofia/external/+4921653258007(a)qsc.de) State Change CS_REPORTING -> CS_DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007(a)qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1380 Session 3 (sofia/external/+4921653258007(a)qsc.de) Locked, Waiting on external entities 2012-01-20 17:01:04.684730 [NOTICE] switch_core_session.c:1398 Session 3 (sofia/external/+4921653258007(a)qsc.de) Ended 2012-01-20 17:01:04.684730 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/+4921653258007(a)qsc.de [CS_DESTROY] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:491 (sofia/external/+4921653258007(a)qsc.de) Callstate Change HANGUP -> DOWN 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:494 (sofia/external/+4921653258007(a)qsc.de) Running State Change CS_DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:504 (sofia/external/+4921653258007(a)qsc.de) State DESTROY 2012-01-20 17:01:04.684730 [DEBUG] mod_sofia.c:374 sofia/external/+ 4921653258007(a)qsc.de SOFIA DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:86 sofia/external/+4921653258007(a)qsc.de Standard DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:504 (sofia/external/+4921653258007(a)qsc.de) State DESTROY going to sleep

4 8

upnp issue I think
by Henry Huang 23 Jan '12

23 Jan '12

I am running Asterisk on my internal network. And I had port forwarding of 5060 from public IP to my Asterisk's internal IP address. I have recently installed a few FreeSWITCH from git on remote locations. And the weird thing is whenever I am trying to register to those FreeSWITCH servers with a softphone on my laptop, the returning SIP message end up routed to the Asterisk instead of my laptop since my softphone default SIP port is 5060. But the same wouldn't happen if I am registering to remote Asterisk servers. The return SIP message will routed correctly back to my softphone. So in summary, I can register to remote Asterisk server even when I've already forwarded port 5060 to my internal Asterisk server, but I can't register to remote FreeSWITCH servers. Does anyone know the reason behind this? Henry

1 0

Cordless IP Phone
by ocset 22 Jan '12

22 Jan '12

Hi I have had great success using Yealink phones with Freeswitch and a customer has asked me for a cordless phone for their office. I want to ensure that they can keep as much functionality with a cordless phone as they have with the Yealink T28. I am looking at the Siemens C610IP phone but don't know how well it plays with Freeswitch. Can someone please shed some light on the Siemens phone or any alternative that you have successfully implemented. Thanks in advance

11 12