September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Asterisk registration with FS
by Noah Silverman 25 Sep '08

25 Sep '08

Hi, I have our FS server configured and running. I can register a polycom phone to it perfectly. Can make calls out and receive calls in. Works well, quality is solid, etc. Now, I want to register an asterisk box with FS. (Our eventual plan is to resell VOIP as a "trunk" to end users, so this is an important test.) FS won't route the calls from asterisk. I don't see any specific errors in the FS console, but the calls just die. Below is the result of "sofia status profile default" showing that both the asterisk box and polycom phone did successfully register. (username, password, host changed for privacy) Call-ID 456fa3406d67ee337c6c81264932f76a(a)127.0.0.1 User 3235551212(a)111.111.111.111 Contact "user" <sip: 3235551212@222.222.222.222:1024;fs_nat=yes> Agent Asterisk PBX Status Registered(UDP-NAT)(unknown) EXP(2008-09-22 12:17:40) Call-ID 8c5f3b02-c1f07a54-7d8ac0c7(a)10.0.1.110 User 3235551212(a)111.111.111.111 Contact "user" <sip: 3235551212@222.222.222.222:5060;fs_nat=yes> Agent PolycomSoundPointIP-SPIP_500-UA/2.1.3.0028 Status Registered(UDP-NAT)(unknown) EXP(2008-09-22 13:57:15) Below is the config in my sip.conf for asterisk. (IP and DID changed for privacy) [Freeswitch] host=111.111.111.111 username=3235551212 secret=password port=5060 type=peer trustrpid=yes sendrpid=yes context=from-trunk canreinvite=no disallow=all allow=ulaw Can anyone help me figure out what is wrong? Thanks, -N

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 25 Sep '08

25 Sep '08

Anthony, this is a simplified version dialplan, just for an illustration of the previous posts: file default.xml: <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="leg=a"/> <action application="set" data="destination=user/$1(a)10.3.1.11"/> <action application="export" data="execute_on_answer=execute_extension ONANSWER XML default"/> <action application="bridge" data=" ="[NEWVARIABLE=AAABBBCCC]${destination}"/> </condition> </extension> <extension name="Call_Answered"> <condition field="destination_number" expression="^ONANSWER$"> <action application="execute_extension" data="INFO XML info"/> </condition> </extension> </context> </include> file info.xml: <xml version="1.0" encoding="utf-8"?> <include> <context name="info"> <extension name="Call_Info"> <condition field="destination_number" expression="^INFO$" /> </condition> <condition field="${leg}" expression="^a$">  <anti-action application="set" data="On_Answer_Is_It_Then_Passed=${NEWVARIABLE}"/> <anti-action application="info"/> </condition> </extension> </context> </include> Sorry about my inconsistent use of the terms, the XML about should explain. /Jon

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IVR Questions
by Jim Flowers 25 Sep '08

25 Sep '08

I'm ready to tackle creating a multi-level IVR. From what I have seen in the wiki, what I have to do is to create a separate wav file for each prompt in some other software and then create an XML file to organize the logic around pointers to the wav files and terminating in the requited actions. Am I missing something or is there any software that would simplify this effort that is worth investigating? -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 25 Sep '08

25 Sep '08

Anthony, I think I understand this part of it, but I'll try to stay humble. This is what I want: I want to have several destinations in my dialstring, and I want to pass different variables on to the script which runs after answer. I also wanted to control the A-number caller ID, and that works as it should, by placing [originate_caller_id_number=XXXX] in front of the destination. So let's forget about that. I have tried: <action application="bridge" data="[NEWVARIABLE=AAABBBCCC]user/${custid}${dialed_ext}@${caller_domain }"/> In order to pass the variable NEWVARIABLE on to the ONANSWER script. I have let the ONANSWER XML script make an "execute_extension call" to another XML script. For the sake of clarity, lets call it the INFO script. In the INFO script I check if variable in the dial string has been given a value. The ONANSWER script is fired off two times: when the a-leg answers and when the b-leg answers. Apparently that's the way "execute_on_answer" works, when its set using the export facility. When it is the a-leg which fires off the ONANSWER script, the NEWVARIABLE exists in the following INFO script fired off by the b-leg-answer event. But if the b-leg which fires off the ONANSWER script first, the variable has no value in INFO script. In some tests, I get a populated variable in 3 out of 10 calls. Hope this clarified the issue, if not, I may call you on the phone. /Jon

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 25 Sep '08

25 Sep '08

Thanks Anthony. I have been confused because of the semantics: execute_on_anwer implies that the code pointed at is re-executed. Anyhow, including the application execute_extension in the ONANSWER-code, helps but not every time: In 20-30% of all the instances, the variable is passed on to the b-leg part of the ONANSWER. In other cases the variable is still undefined. When the b-leg of the ONANSWER runs before the a-leg part, it is undefined, and vice versa. The code executed by execute_extension is: <?xml version="1.0" encoding="utf-8"?> <include> <context name="onanswer"> <extension name="Call_Answered"> <condition field="destination_number" expression="^ONANSWER$"> </condition> <condition field="${leg}" expression="^a$"> <action application="info"/>  <anti-action application="set" data="On_Answer_Is_It_Then_Passed=${NEWVARIABLE}"/> <anti-action application="info"/> </condition> </extension> </context> </include> Jon

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Reading DTMF via Event Socket
by Klaus Teller 25 Sep '08

25 Sep '08

Hi, Please help: 1) How do i send DTMF via the socket events interface? 2) It seems playing audios (playback) is non-blocking. Is this the intended behavior? How can i make it blocking? Alternately, how can i know when the audio is completed? Thanks, Klaus. -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

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Voicemail x DID
by Jair Santos 25 Sep '08

25 Sep '08

Hi, If I call ext 1000 the voicemail system answer on timeout . If I call a DID that is linked to that same extension it returns a busy signal when it is trying to call the VM. In my public.xml I have <extension name="public_did"> <condition field="destination_number" expression="^(3462101)$"/> <condition field="${sip_h_X-DID}" expression="^(3105266066)$"> <action application="transfer" data="1000"/> <action application="voicemail" data="default $${domain} 1000"/> </condition> </extension> and the error is 2008-09-24 14:43:08 [NOTICE] mod_sofia.c:1116 sofia_receive_message() Pre-Answer <mailto:sofia/external/5888606894@216.143.130.65> sofia/external/5888606894(a)216.143.130.65! 2008-09-24 14:43:23 [INFO] switch_rtp.c:1253 rtp_common_read() Auto Changing port from 216.143.130.65:12622 to 192.168.1.117:31580 2008-09-24 14:43:38 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup <mailto:sofia/external/5888606894@216.143.130.65> sofia/external/5888606894(a)216.143.130.65 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2008-09-24 14:43:38 [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening File [/usr/local/freeswitch/storage/voicemail/default/voipclic.com/1000/msg_60245 7c9-7ec8-4d93-a233-6e2a51b33099.wav] [System error : No such file or directory.] 2008-09-24 14:43:38 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 1 ( <mailto:sofia/external/5888606894@216.143.130.65> sofia/external/5888606894(a)216.143.130.65) Ended 2008-09-24 14:43:38 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel <mailto:sofia/external/5888606894@216.143.130.65> sofia/external/5888606894(a)216.143.130.65 [CS_HANGUP] Any help will be appreciated. thanks Jair Santos

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LDAP for user authentication & phone numbers
by Adrian Gschwend 25 Sep '08

25 Sep '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi group, I know this topic comes up every few months so sorry for bringing it up again ;) I use FreeSWITCH for our open source project for quite some time now and we are very happy with it. To manage users I just use LDAP and every application (Wiki, TRAC, Apache, XMPP...) is connecting to the directory for users & authentication. So I would like to do the same for FreeSWITCH. Googling a bit around I found some old postings on this mailinglist, namely the one from Anthony where he explains a bit about mod_ldap: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003382.html Quoting that mail: "Now also in mod_ldap, there is some code someone recently contributed to tie our XML interface to LDAP, This is more interesting because then when anything in FreeSWITCH tries to lookup a user, dialplan entry or anything else in our XML config, all the important details are passed to LDAP where it can make a query, pull the info out of LDAP and deliver it back to FreeSWITCH as the XML it was looking for. This allows you to make all of the registrations, dialplan etc real time driven by LDAP, you can also bind a perl or lua script to this operation as well as mod_xml_curl who will turn the request into an HTTP post to a web server to fetch the data." That's basically what I need. My idea is to not create an XML file for every user but to tell (in an XML file I guess) that the phone numbers, username & password can be found in a directory. I do have quite some experience in LDAP debugging and I can code a bit as long as I don't have to mess with pointers ;) So where should I study some code to figure out how that works exactly? I promise to document it once I got it to work as I think more people want to have that feature as well. thanks Adrian - -- Adrian Gschwend @ netlabs.org ktk [a t] netlabs.org - ------- Open Source Project http://www.netlabs.org -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFI25cbqpMUYrZbQBERAuAgAJ0bFoa7HHCqDB/+ORStts/4GcGnngCeIt8o No7BU937l+OOkWiT/qzRhjE= =np/p -----END PGP SIGNATURE-----

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge
by Jon Bruel 25 Sep '08

25 Sep '08

I have made some further testing, and even without an "execute_on_answer" in the ONANSWER-code, the variable set in the dialstring appears now and then. And it only happens, when the ONANSWER code for the a-leg is executed first. On my system, this happens in 30-50% of the tests. If I move the ONANSWER code to another file, where this is the only condition, it does not work, but entering some dummy conditions before, makes it work now and then. So this may be a concurrency issue? Jon

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working PHP script with socket
by Gopal krishnan 25 Sep '08

25 Sep '08

Hi, Outbound call went thru socket with the PHP program. The program as follows, <?php $Caller = htmlspecialchars($_POST['Caller']); $Callee = htmlspecialchars($_POST['Callee']); echo "Php now! "; echo "Caller is: $Caller "; echo "Callee is: $Callee "; $handle = fsockopen('127.0.0.1', '8021', $erno, $errstr, 30); if (!$handle) { //if connection fails exit and tell us what went wrong. die("Connect failed with erno $erno and errstr $errstr\n"); } else { //read the first line the smtp server gives us (usually the version string) //$responses = fgets($handle, 1024); $response= fread($handle, 1024); echo "connection opened, got response: $response "; $authMessage = "auth ClueCon\r\n\r\n"; fwrite($handle, "$authMessage"); $response = fread($handle, 1024); echo "authMessage sent, got response: $response "; $command = "api originate sofia/default/$Callee@<freeswitch IP address> $Caller \r\n\r\n"; echo "command is $command"; fwrite($handle, $command); $response = fread($handle, 1024); echo "command sent, got response: $response "; $response = fread($handle, 1024); echo "command sent, got response: $response "; //fclose($handle); } } ?> Thanks -- Thank you with regards, Gopal,

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