November 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH suddenly stopped accepting calls
by Hector Geraldino 30 Nov '12

30 Nov '12

Hello everyone, I have a FreeSWITCH instance that has been running for months without an issue, until last Monday. The setup is for a hotline where we receive incoming calls transferred from the Client Cisco UCM to our FreeSWITCH instance, and the calls are handled by a custom Java application via ESL. As I said it was working flawlessly for weeks, but last Monday FreeSWITCH just stopped accepting new calls. I was able to see the SIP messages flowing back and forth from FS<->UCM, but looks like FreeSWITCH was unable to create new channels. Once I restarted FreeSWITCH everything was back to normal. Here are links to the SIP/console logs I was able to capture which covers both scenarios (non-working FS and working FS): Non-working: http://pastebin.freeswitch.org/20275 Working: http://pastebin.freeswitch.org/20274 As you can see, in the non-working scenario no channels were created on FreeSWITCH, so no 200 ACK message was sent back to UCM after the INVITE, which caused the UCM to send a CANCEL message after a few seconds and a 487 Request terminated from FS -> UCM. Restarting the application fixed the issue, but I have a couple of questions regarding this: 1) We have a lot of monitoring in place for all our applications, and I could probably alerted our Operations team if FreeSWITCH crashed or some error was logged in the FS log. However I was unable to found anything meaningful in the logs that indicates any sort of error. It just suddenly stopped creating new channels, and we get the alert/complain from the client - ouch! 2) I was also not sure what other information I should collect when something like this happens. As the application didn't crash it didn't create any coredump files. I know that restarting it fixes the problem, but that is not what the client wants to hear. Any recommendations on what to do/look at to proactively detect this kind of issues and/or how to collect the appropriate info to get the root cause will be greatly appreciated. Thank you all for your time, Hector

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Routing call to another interface
by William Alianto 30 Nov '12

30 Nov '12

Hi, I got a server with this configuration : Gateway ^ | | | | eth 2 (10.x.x.x) | | FS<-------->eth1 (180.x.x.x) <------->Internet<---------->User Now I need to route the call to/from gateway. Would anybody help me to configure a dialplan for this problem? Regards

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Compilation Issue
by Gilbert T. Gutierrez, Jr. 30 Nov '12

30 Nov '12

I am having an issue compiling 1.2 Stable this morning. My notes are all below. When I compile a second time, it appears to compile ok. I am not sure if I trust that compilation because some of the modules may not be complete. What do you guys feel the problem may be? Gilbert Centos6 64bit The procedure I am following: cd /usr/src git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git cd freeswitch ./bootstrap.sh -j ./configure --without-libcurl -C make -j `cat /proc/cpuinfo |grep processor |wc -l` The make first throws the following error... "In file included from ./src/include/private/switch_core_pvt.h:44, from src/switch.c:48: /usr/src/freeswitch/libs/apr/include/apr_pools.h:37:17: error: apr.h: No such file or directory" Everything following is an error terminating in "make: *** [all] Error 2"

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Brief show notes for today in the world of FreeSWITCH <PLEASE READ ME>
by Ken Rice 30 Nov '12

30 Nov '12

Hey Guys, Coming up today FreeSWITCH 1.2.5.2 this is another minor release to catch a few regressions we missed in 1.2.5.1. Yes we are aware of them and this will be released sometime today, please watch the mailing list and other ways you get your FS news. Also Don¹t forget today is Friday, so the Friday FreeForAll will be in Full Effect today. Join us at sip:888@conference.freeswitch.org and see what the FreeForAll is all about And speaking of 888, 9888 in your Default Example Config FreeSWITCH installs is one way to get to the FreeSWITCH conference bridge. While we do appreciate that you may want to call 888 on occation to test, please refrain from calling 888 with 10 or 20 concurrent calls this is a public conference bridge for the community and spamming it only makes is less available for others to use. K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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GEMopen and freeswitch for voice call and sms
by hala alramli 30 Nov '12

30 Nov '12

Hello all, i want to ask if Huawei E355 and E369 dongle compatible with freeswich and GSMopen for SMS and voice calls. thanks

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[Confidential] - uuid_broadcast
by Nuwan Wijerathne 30 Nov '12

30 Nov '12

Hi Anthony, I have attached the log file to the JIRA case (FS-4884) as you requested . Please let me know if you need any more information. Thank you, Nuwan. This e-mail and any attachments are for the intended addressee(s) only and may contain confidential and/or privileged material. If you are not a named addressee, do not use, retain or disclose such information. This email is not guaranteed to be free from viruses and does not bind Unify in any contract or obligation. Unify Business Solutions Ltd. Registered in England and Wales. No: 4749638 Registered Office: Ambassador House, 5 Midland Way, Barlborough, S43 4XA United Kingdom.

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About the display problem during calling
by sparklezou 30 Nov '12

30 Nov '12

Hi All, I facing a strange problem. Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine. But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number. I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem. I checked the sip info. The "Outbound Call" is in the 200 SDP message. the called is set to ""Remote-Part-ID: "Outbound Call" <sip: 1234(a)a.b.c.d>; ....... " Not the correct called name. Is there anything wrong? Please help me. Thanks! 2012-11-30 sparklezou

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Trouble with voicemail say digits
by Komar, Jason 30 Nov '12

30 Nov '12

Just updated to FS 1.2.5.1 on Gentoo from 1.2.3. I have one extension that uses the default voicemail greeting. Since I updated, when it gets to the part where it says the extension number, it cannot find the wav files for the digits. It is looking for them in the digits folder and not in the frequency subfolders (i.e. 16000 32000 48000 8000). As soon as it hits this error, the voicemail app says goodbye and hangs up. 2012-11-28 20:35:17.068585 [DEBUG] switch_ivr_play_say.c:244 Handle say:[2003] (en:en) 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File [/opt/freeswitch/sounds/en/us/callie/digits/2.wav] does not exist. 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist. 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File [/opt/freeswitch/sounds/en/us/callie/digits/0.wav] does not exist. 2012-11-28 20:35:17.068585 [ERR] mod_dptools.c:4482 File [/opt/freeswitch/sounds/en/us/callie/digits/3.wav] does not exist. Is there a config option to specify the frequency subfolder, or did something not build correctly? Thanks, Jason Komar

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sqlite3 and regex
by Scott 30 Nov '12

30 Nov '12

"LIKE" notwithstanding, sqlite3 does not have a built-in true regex function; it does allow for a a c-language hook to one. Given fs extensive use of the regex engine and of sqlite3, we're wondering if the hook is already written and rolled. If so, can the rest of us hook it to our sqlite3 uses (e.g. from dial plan lua sqlite3). With thanks,

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Problem accessing channel variables
by Richard Gration 30 Nov '12

30 Nov '12

Hi all, I'm trying to write something to fit into a two factor auth mechanism. I have a CGI script which is called with a phone number and a PIN as parameters. In the script I use the event socket api to place a call. The PHP code I'm using to write to the socket is straight out of the FS docs (can't remember exactly, I think I got it from the wiki). This is my api command: api originate {originate_timeout=30}sofia/external/${dExt}@MGC '&play_and_get_digits(4 4 2 30000 # torvalds-says-linux.wav torvalds-says-linux.wav foo \d+)' This all works fine, up to a point. The call is made, the wav is played. But then PAGD is supposed to put the digits into a channel var, in this case "foo". But I've been having trouble getting the value of this channel var *reliably*. If the # key is used to terminate the input, then I can get the channel var value with api uuid_getvar $uuid foo However, if the input is terminated by PAGD because 4 digits have been entered, then the call is hung up (by FS/PAGD) before I have a chance to get the value of the channel var. Can anyone help here please? I suppose I can work around this by increasing the digits to 5 and telling people to use # to terminate the input, but I'd like to understand the correct way to use PAGD if possible. If you think I'm going about this the wrong way entirely, then please suggest an alternative :-) My code is below. As I say, lines 1-71 are straight out of the docs. 1 <?php 2 3 $password = "xxxxxx"; 4 $port = "xxxx"; 5 $host = "127.0.0.1"; 6 7 function event_socket_create($host, $port, $password) { 8 $fp = fsockopen($host, $port, $errno, $errdesc) 9 or die("Connection to $host failed"); 10 socket_set_blocking($fp,false); 11 12 if ($fp) { 13 while (!feof($fp)) { 14 $buffer = fgets($fp, 1024); 15 usleep(100); //allow time for reponse 16 if (trim($buffer) == "Content-Type: auth/request") { 17 fputs($fp, "auth $password\n\n"); 18 break; 19 } 20 } 21 return $fp; 22 } 23 else { 24 return false; 25 } 26 } 27 28 function event_socket_request($fp, $cmd) { 29 30 if ($fp) { 31 fputs($fp, $cmd."\n\n"); 32 usleep(200); //allow time for reponse 33 34 $response = ""; 35 $i = 0; 36 $contentlength = 0; 37 while (!feof($fp)) { 38 $buffer = fgets($fp, 4096); 39 if ($contentlength > 0) { 40 $response .= $buffer; 41 } 42 43 if ($contentlength == 0) { //if contentlenght is already don't process again 44 if (strlen(trim($buffer)) > 0) { //run only if buffer has content 45 $temparray = split(":", trim($buffer)); 46 if ($temparray[0] == "Content-Length") { 47 $contentlength = trim($temparray[1]); 48 } 49 } 50 } 51 52 usleep(200); //allow time for reponse 53 54 //optional because of script timeout //don't let while loop become endless 55 if ($i > 90000) { break; } 56 57 if ($contentlength > 0) { //is contentlength set 58 //stop reading if all content has been read. 59 if (strlen($response) >= $contentlength) { 60 break; 61 } 62 } 63 $i++; 64 } 65 66 return $response; 67 } 68 else { 69 echo "no handle"; 70 } 71 } 72 73 function strip_cr($str) { 74 return preg_replace("/\n/",'',$str); 75 } 76 77 $dExt = $_GET['dExt']; 78 if ($dExt) { 79 $fp = event_socket_create($host, $port, $password); 80 81 # Place call and find uuid 82 $cmd = "api originate {originate_timeout=30}sofia/external/677${dExt}(a)194.145.191.131 '&play_and_get_digits(4 4 2 30000 # torvalds-says-linux.wav torvalds-says-linux.wav foo \d+)'"; 83 $response = event_socket_request($fp, $cmd); 84 $response = preg_replace('/</','<',$response); 85 $response = preg_replace('/>/','>',$response); 86 $response = strip_cr($response); 87 $bits = preg_split('/\s+/',$response); 88 $uuid = $bits[1]; 89 90 echo '<pre>'; 91 echo date('H:i:s') . "\n"; 92 echo "call placed: $cmd\n"; 93 echo "response :$response:\n"; 94 echo "uuid $uuid\n"; 95 96 # Retrieve channel variable 97 $cmd = "api uuid_getvar $uuid foo"; 98 $response = ''; 99 while (! $response or $response == '_undef_') { 100 echo "in loop : got _undef_\n"; 101 $response = event_socket_request($fp, $cmd); 102 $response = strip_cr($response); 103 } 104 echo "response :$response:\n"; 105 echo '</pre>'; 106 107 # Close socket 108 fclose($fp); 109 } else { 110 echo 'Request URL in this format: /test.php?dExt=<phone_number>' . "\n"; 111 } 112 113 ?> Cheers, Rich -- Once our basic material needs are met - in my utopia, anyway - life becomes a perpetual celebration in which everyone has a talent to contribute. But we cannot levitate ourselves into that blessed condition by wishing it. We need to brace ourselves for a struggle against terrifying obstacles, both of our own making and imposed by the natural world. And the first step is to recover from the delusion that is positive thinking. -- Barbara Ehrenreich

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