July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Answer issue on inbound call
by Max Alex 15 Sep '11

15 Sep '11

Hi, I have installed freeswitch latest version with sangoma card A200 as well, I have installed and configured freetdm module with wanpipe drivers for freeswitch, We are properly receiving the inbound calls in public context and then we are routing that call to 1001 extension, it is properly routing to 1001 as well, but we have one issue while routing on 1001. Here is the issue description. I am calling from my cell phone to that DID number of pri line, and then it will start ringing on 1001 extension, When 1001 extension start ringing the call is answered on my cell phone, it is something like freeswitch preanswer or autoanswer the call, how can i stop this answer call when it is ringing on 1001 extension, Waiting for good reply. Thanks, Max Alex

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ESL Send MESSAGE_WAITING Error - Cannot find profile
by Avi Marcus 13 Sep '11

13 Sep '11

I'm trying to send a MESSAGE_WAITING via ESL on multi-tenant, but I'm getting this error: 2011-06-27 10:46:25.443081 [ERR] sofia_presence.c:405 Cannot find profile 20064.domain.com] the cli lists the domain as an alias to the internal profile. what should I do? I am writing some test code in order to try to send MWI events using the ESL Manager code DLL. Here is my test code: static void InboundMode2(Object stateInfo) { //Initializes a new instance of ESLconnection, and connects to the host $host on the port $port, and supplies $password to freeswitch ESLconnection eslConnection = new ESLconnection(myinfo); if (eslConnection.Connected() != ESL_SUCCESS) { Console.WriteLine("Error connecting to FreeSwitch"); return; } //Set log level //ESL.eslSetLogLevel((int)enLogLevel.DEBUG); eslConnection.Api("reloadxml", string.Empty); // Subscribe to all events ESLevent eslEvent2 = eslConnection.SendRecv("event plain ALL"); if (eslEvent2 == null) { Console.WriteLine("Error subscribing to all events"); return; } ESLevent eslEvent = new ESLevent("MESSAGE_WAITING", null); eslEvent.AddHeader("MWI-Messages-Waiting", "yes"); eslEvent.AddHeader("MWI-Message-Account", "103(a)20064.domain.com<103(a)20064.cmvtesttele.com> "); eslEvent.AddHeader("MWI-Voice-Message", "1/1 (1/1)"); eslEvent = eslConnection.SendEvent(eslEvent); if (eslEvent == null) { Console.WriteLine("event error"); return; } //Turns an event into colon-separated 'name: value' pairs. The format parameter isn't used Console.WriteLine(eslEvent.Serialize(String.Empty)); // Grab Events until process is killed while (eslConnection.Connected() == ESL_SUCCESS) { eslEvent = eslConnection.RecvEvent(); Console.WriteLine(eslEvent.Serialize(String.Empty)); } } When I send the event, I am getting a debug message from FS as follows: 2011-06-27 10:46:25.443081 [ERR] sofia_presence.c:405 Cannot find profile 20064.domain.com] Thanks! -Avi

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intermittent EXCHANGE_ROUTING_ERROR
by Bill W. 11 Sep '11

11 Sep '11

Hey All, About once or twice a day, I'm getting an EXCHANGE_ROUTING_ERROR failure. I've enabled debugging, including logging SIP messages, and can't seem to find the cause of the error. Freeswitch goes from setting variables to exchange routing error: switch_channel.c:935 EXPORTING[export_vars] to switch_ivr_originate.c:2447 Cannot create outgoing channel of type [sofia] cause: [EXCHANGE_ROUTING_ERROR] Whenever that happens, "switch_channel.c:816 New Channel" is never called and I don't see any outbound SIP messages to the IP in question. So if no new channel is created, how can there be a routing error? Any insight or advice on how to debug this would be greatly appreciated. Thanks!

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FS to a Sonus SIP trunk
by michael knop 11 Sep '11

11 Sep '11

Hi all! I’m trying to connect my FS to a Sonus SIP trunk. I followed the instruction at http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus but it did not work. At the beginning of a call voice quality is good. After a while it changes to choppy. I don’t know if it’s the same problem: When I call the Tetris extension via Sonus SIP trunk the sound is too fast and I’m getting log entries like the following one: [...] 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME not supported, changing our end from 20 to 10 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from PCMA@20ms@8000hz to PCMA@10ms@8000hz 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global timer resolution to 10ms to handle interval 10 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer [soft] 80 bytes per 10ms 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec sofia/external/+4940...@193...:5060 PCMA/8000 10 ms 80 samples 64000 bits 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write Buffer at 160 bytes to accommodate 320->160 [...] This problem is fixed by adding the following line to conf/sip_profiles/external.xml: <param name="rtp-autofix-timing" value="false"/> Any hints? /micha

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Can't make mod_rtmp
by peely 07 Sep '11

07 Sep '11

Hi, I'm having trouble building FreeSWITCH with the latest mod_rtmp goodness. On Ubuntu X64 compiling with gcc I get: cc1: warnings being treated as errors /usr/src/freeswitch/src/mod/endpoints/mod_rtmp/mod_rtmp.c: In function ‘rtmp_function’: /usr/src/freeswitch/src/mod/endpoints/mod_rtmp/mod_rtmp.c:1612:114: error: comparison between ‘switch_call_cause_t’ and ‘enum <anonymous>’ I'm not sure if I need to do anything else, I just naively added endpoints/mod_rtmp to modules.conf and did a make clean install? Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229… Sent from the freeswitch-users mailing list archive at Nabble.com.

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freetdm api - ftdm_channel_read()
by Juraj Fabo 29 Aug '11

29 Aug '11

Hi I'm prototyping an application which uses libfreetdm above the sangoma isdn stack and sangoma a104d card. I would like to ask for help about proper reading of voice from the particular channel. According to return value of sangoma_get_rx_queue_sz() the default rx queue size is 10. I did tests when application skipped several reads on given channel, then called ftdm_channel_read() w/ zero timeout in a loop while the return value of read was successfull - it was really 10 frames which were then possible to read. How can an application do this reading of a channel by the means of the freetdm api in more nice way? I did not find a better way yet, than to read it in the loop. Any way to explicitly read the number of available frames? Actually, why the ftdm_channel_read() does not read them ALL at once and sets the number of read bytes via *datalen output parameter? (assuming provided *data buffer is large enough) In my tests, always 160 bytes were returned in one particular read. thank you juraj

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V92 modem with freeswitch
by ankIT WALiA 24 Aug '11

24 Aug '11

Hi all, I found a V.92 56k PCI Modem in my old hardware. I want to run some basic functionality test with my PSTN phone line and FS. Can I connect my phone line with FS using this card. If yes how do we configure to use this card. I am new to telephony. I don't know if whatever I said above make sense. Thanks Ankit

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How to authencticate with mod_rtmp?
by Dmitry Kravchenko 23 Aug '11

23 Aug '11

Hi! It is said in book that user registry is single and centralized for all components (p. 64). But while trying to login via FLEX telephone to default number of 1000 with password of 1234, I get "Authentication failed" message. In freeswitch CLI I see message 2011-07-19 15:46:51.473828 [WARNING] rtmp.c:227 Authentication failed for 1000(a)192.168.10.196 where 192.168.10.196 is true IP of my freeswitch machine. Connection with X-Lite goes OK with message 2011-07-19 15:50:19.473829 [WARNING] sofia_reg.c:1337 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000(a)192.168.10.196] from ip 192.168.10.56 If I put wrong password into X-Lite then I get 2011-07-19 15:50:19.493774 [WARNING] sofia_reg.c:1295 SIP auth failure (REGISTER) on sofia profile 'internal' for [1000(a)192.168.10.196] from ip 192.168.10.56 So my conclusion is that SIP phone is treates as internal profile by default. Is it possible to set RTMP profiles as internal too? Or may be I should better create exteral accounts to authenticate with RTMP? Thanks.

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vm-alternate-greet-id is ignored
by Sascha Daniels 22 Aug '11

22 Aug '11

Hi together, I need to set a different Number in voicemail greetings. Just for testing purpose I tried to set it without a variable. xml.condition( :field => 'destination_number', :expression => '^-vbox-(.+)$' ) { xml.action( :application => 'answer', :data => 'voicemail_authorized=true' ) xml.param( :name => 'vm-alternate-greet-id', :value => '4444' ) xml.action( :application => 'voicemail', :data => 'default ${domain_name} $1' ) } Unfortunately the real sip account is played. Did I get the documentation wrong? I am using "FreeSWITCH version: 1.0.head (git-2e651c8 2011-07-03 22-35-44 -0500)" Regards Sascha -- AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied --> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister Montabaur B14998 Bücher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de

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Mod_rad_auth issue for FS working with FreeRadius server
by fieldpeak 14 Aug '11

14 Aug '11

Hello Gurus, i met a issue when using mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works with freeradius server+mysql for AAA, the details is below, Could anyone give any hints, Thanks in advance. i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however, when i dialed 601 to trigger the dial plan, the console show errors, it looks "h323-conf-id" is not in the directory, then i tried to add this attribute to the dictionary, however, it does not help, in the wiki, it mentioned the rad_auth.conf.xml contains <param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did not find the file "dictionary.all" at that directory, so i use dictionary. BTW, the freeradius server + mysql works well. console errors: EXECUTE sofia/internal/1001(a)124.193.106.104 auth_function(in , in 38516060333, in 003282, out AUTH_RESULT) 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate initial structure. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed configuration. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver := 127.0.0.1:1812:gateway. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary := /usr/local/etc/radiusclient/dictionary. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile := /var/run/radius.seq. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile := /usr/local/etc/radiusclient/port-id-map. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set default_realm := . 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_timeout := 3. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_retries := 2. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_deadtime := 0. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr := *. 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius: User-Name: 38516060333 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius: User-Password: 003282 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius: Called-station-Id is empty, ignoring... 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle attribute: h323-conf-id 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not found in dictionary 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending radius packet. 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured during RADIUS Authentication(RC=-1) 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured during radius authorization. EXECUTE sofia/internal/1001(a)124.193.106.104 log(INFO AUTH_RESULT=) 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202 AUTH_RESULT= EXECUTE sofia/internal/1001(a)124.193.106.104 log(INFO billing_model=) 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202 billing_model= EXECUTE sofia/internal/1001(a)124.193.106.104 log(INFO credit_amount=) 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202 credit_amount= EXECUTE sofia/internal/1001(a)124.193.106.104 log(INFO currency=) 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202 currency= EXECUTE sofia/internal/1001(a)124.193.106.104 log(INFO preffered_lang=) 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202 preffered_lang= added below in the dictionary(/usr/local/etc/radiusclient/dictionary): ATTRIBUTE h323-conf-id 1008 string dial plan: <extension name="unitest_rad-ANI-auth"> <condition field="destination_number" expression="^601$"> <action application="log" data="INFO Before Auth "/> <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/> <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/> <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>  <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/> <action inline="true" application="set" data="USERNAME=38516060333"/>  <action inline="true" application="set" data="PASSWD=003282"/>  <action application="sleep" data="2000"/> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/> <action application="log" data="INFO billing_model=${billing_model}"/> <action application="log" data="INFO credit_amount=${credit_amount}"/> <action application="log" data="INFO currency=${currency}"/> <action application="log" data="INFO preffered_lang=${preffered_lang}"/> <action application="log" data="INFO credit_time=${credit_time}"/> <action application="log" data="INFO h323_ivr_duration=${h323_ivr_duration}"/> <action application="log" data="INFO return_code=${return_code}"/>  </condition> </extension> radius_cdr.conf.xml: <configuration name="radius_cdr.conf" description="RADIUS CDR Configuration"> <settings>  <param name="dictionary" value="/usr/local/freeswitch/conf/radius/dictionary"/>  <param name="radius_retries" value="3"/>  <param name="radius_timeout" value="5"/>    <param name="acct_server" value="127.0.0.1:1813:testing123"/> </settings> </configuration> the FS version: FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400) Regards, Charles

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