January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch as gateway
by Nikhil 24 Jan '12

24 Jan '12

Hi I would like to use freeswitch as a Voip Gate way like Analog,ISDN,GSM. Is this possible? if yes provide some hint. Thanks

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missing digit from destination number, overlap dialing
by Andreas Gabler, ABX 24 Jan '12

24 Jan '12

Dear all, May I ask you for help? We use Freeswitch with Sangoma B700 FlexBRI on Windows XP professional SP3. We have 2 BRI lines. Some external calls lack the last digit of the destination number. Many destination numbers are ok. It depends from the source of the call. Calls from German mobile network have complete number. Many calls from German landline have complete number. A few calls from German landline lack the last digit. This callers hear "Called number is not available". Freeswitch received the complete number from 68% of the calling numbers at all times. This means 7 digits after the country code. Incomplete number from 14% at all times. Only 6 digits. 18% varied. Sometimes we received 6 digits. Sometimes 7. I tried <param name="overlap" value="yes"/> in the file freetdm.conf.xml. This makes the external caller do not receive an answer. He hears nothing. ---------------------------------------------------------------------- This is an extraction of the freeswitch.log without overlap=yes: Line numbers 02, 04 and 19 lack the last digit. Line numbers 01, 03 and 10 from mobile network. Line number 13 anonymous caller. Other from landline. 01 incoming call: called no:[4166065] calling no:[1722051138] 02 incoming call: called no:[416604] calling no:[3517969878] 03 incoming call: called no:[4166030] calling no:[1738920856] 04 incoming call: called no:[416603] calling no:[6423969575] 05 incoming call: called no:[4166033] calling no:[6423969575] 06 incoming call: called no:[4166041] calling no:[35284193228] 07 incoming call: called no:[4166038] calling no:[970891000] 08 incoming call: called no:[4166012] calling no:[3514129670] 09 incoming call: called no:[4166012] calling no:[35284193212] 10 incoming call: called no:[4166054] calling no:[1733583343] 11 incoming call: called no:[4166052] calling no:[35284153322] 12 incoming call: called no:[4166021] calling no:[3571608532] 13 incoming call: called no:[4166020] calling no:[] 14 incoming call: called no:[4166030] calling no:[2518347351] 15 incoming call: called no:[4166033] calling no:[352371791] 16 incoming call: called no:[4166054] calling no:[35284595951] 17 incoming call: called no:[4166030] calling no:[2518347351] 18 incoming call: called no:[4166019] calling no:[35183142120] 19 incoming call: called no:[416602] calling no:[351828240] 20 incoming call: called no:[4166010] calling no:[7022609250] ---------------------------------------------------------------------- This is a part of freeswitch.log without overlap=yes: Calling from 35284041831 to 4166069 is ok. 07:59:04.835904 [INFO] ftmod_sangoma_isdn_stack_hndl.c:142 [s2c1][2:1] Incoming call: Called No:[4166069] Calling No:[35284041831] 07:59:04.835904 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:199 [s2c1][2:1] Changed state from DOWN to RING 07:59:04.835904 [DEBUG] ftdm_state.c:511 [s2c1][2:1] Executing state processor for RING 07:59:04.835904 [DEBUG] ftmod_sangoma_isdn.c:623 [s2c1][2:1] processing state change to RING 07:59:04.835904 [DEBUG] ftmod_sangoma_isdn.c:647 [s2c1][2:1] Sending incoming call from 35284041831 to 4166069 to FTDM core 07:59:04.835904 [DEBUG] ftmod_sangoma_isdn.c:884 [s2c1][2:1] Completed state change from DOWN to RING in 1ms ... ---------------------------------------------------------------------- This is a part of freeswitch.log with overlap=yes: Calling from 35284041728 to 4166065 failed. 10:17:43.696875 [INFO] ftmod_sangoma_isdn_stack_hndl.c:142 [s2c1][2:1] Incoming call: Called No:[4166065] Calling No:[35284041728] 10:17:43.696875 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:197 [s2c1][2:1] Changed state from DOWN to COLLECT 10:17:43.696875 [DEBUG] ftdm_state.c:511 [s2c1][2:1] Executing state processor for COLLECT 10:17:43.696875 [DEBUG] ftmod_sangoma_isdn.c:623 [s2c1][2:1] processing state change to COLLECT 10:17:43.696875 [INFO] ftmod_sangoma_isdn_stack_out.c:102 [s2c1][2:1] Sending SETUP ACK (suId:1 suInstId:5 spInstId:5 dchan:2 ces:0) 10:17:43.696875 [DEBUG] ftmod_sangoma_isdn.c:884 [s2c1][2:1] Completed state change from DOWN to COLLECT in 0ms 10:17:55.775000 [INFO] ftmod_sangoma_isdn_stack_rcv.c:172 [s2c1][2:1] Received INFO (suId:1 suInstId:5 spInstId:5 ces:0) 10:17:55.775000 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:368 [s2c1][2:1] Processing INFO (suId:1 suInstId:5 spInstId:5 ces:0) 10:17:55.775000 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:464 [s2c1][2:1] Processing INFO (suId:1 suInstId:5 spInstId:5) 10:18:08.337500 [INFO] ftmod_sangoma_isdn_stack_rcv.c:210 [s2c1][2:1] Received DISCONNECT (suId:1 suInstId:5 spInstId:5) 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:523 [s2c1][2:1] Processing DISCONNECT (suId:1 suInstId:5 spInstId:5) 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:547 [s2c1][2:1] Changed state from COLLECT to CANCEL 10:18:08.337500 [DEBUG] ftdm_state.c:511 [s2c1][2:1] Executing state processor for CANCEL 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:623 [s2c1][2:1] processing state change to CANCEL 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:753 [s2c1][2:1] Hanging up call before informing user! 10:18:08.337500 [INFO] ftmod_sangoma_isdn_stack_out.c:420 [s2c1][2:1] Sending RELEASE/RELEASE COMPLETE (suId:1 suInstId:5 spInstId:5) 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:759 [s2c1][2:1] Completed state change from COLLECT to CANCEL in 0ms 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:759 [s2c1][2:1] Changed state from CANCEL to HANGUP_COMPLETE 10:18:08.337500 [DEBUG] ftdm_state.c:511 [s2c1][2:1] Executing state processor for HANGUP_COMPLETE 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:623 [s2c1][2:1] processing state change to HANGUP_COMPLETE 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:831 [s2c1][2:1] Waiting for release from stack 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:884 [s2c1][2:1] Completed state change from CANCEL to HANGUP_COMPLETE in 1ms ---------------------------------------------------------------------- You can download an attachment (3 MBytes): http://www.abx.de/temp/freeswitch/2012-01-24/attachment.zip The attachment contains dialplan: default.xml default-anrufverteilung-nach-isdn.xml public.xml (Public directory is empty.) config: freeswitch.xml freetdm.conf freetdm.conf.xml Wanpipe.pdf log: full freeswitch.log ---------------------------------------------------------------------- Thank you for help. Best regards Andreas Gabler gabler(a)abx.de ------------------------------------------- ABX advanced biochemical compounds Heinrich-Glaeser-Str. 10-14 D-01454 Radeberg Germany telephone: +49 3528 4041 740 facsimile: +49 3528 4041 8877 email: info(a)abx.de web: www.abx.de ------------------------------------------- general manager: Dr. Peter Moll company headquarter: D-01454 Radeberg VAT number: DE 812 136 673 Commercial Registry: HRB 14041 Amtsgericht Dresden tax authority: Finanzamt 3213 Hoyerswerda tax number: 213/105/00531 ------------------------------------------- Please, use plain text format instead of html in your emails. Thank you. -------------------------------------------

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How to Set Call Priority in mod_callcenter
by amit nakum 24 Jan '12

24 Jan '12

Dear All, I have set queue parameter to <param name="time-base-score" value="system"/>,so now how to raise some caller's score,allows them to receive priority over other normal calls that might have been in the queue longer. I have test above scenario using <param name="time-base-score" value="system"/> which is not fulfill my requirement,if there is any parameter to set this scenario. Pl help me to set this scenario. Thanks and Regards Amit Nakum

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SouthAmerican Users
by Ken Rice 24 Jan '12

24 Jan '12

Hey guys, Maybe one of you can help me out... Looking for someone in South America (Brazil preferably) that has good Colocation Space K

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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

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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

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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,

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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

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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

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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>

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