May 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Error load mod_gsmopen
by tangdu 26 May '11

26 May '11

Hi all, I have followed this wiki page to intall gsmopen module http://wiki.freeswitch.org/wiki/GSMopen#Linux Installed following prerequisites packages # yum -y install alsa-lib-devel alsa-utils I downloaded gsmlib-1.10-12ubuntu1 and patched, compiled it no error. #cd /#/usr/local/src/freeswitch-1.0.7/src/mod/endpoints/mod_gsmopen/gsmlib/gsmlib-1.10-patched-12ubuntu1 #make clean #make install Nokia-3208c connected with USB. Use minicom,I can dialout from GSM. When I load mod_gsmopen, following error message. What's wrong? freeswitch(a)localhost.localdomain> load mod_gsmopen 2011-05-26 15:27:45.465196 [INFO] mod_enum.c:808 ENUM Reloaded 2011-05-26 15:27:45.465196 [INFO] switch_time.c:954 Timezone reloaded 530 definitions 2011-05-26 15:27:45.503006 [WARNING] mod_gsmopen.cpp:1860 rev [(nil)|37 ][WARNINGA 1860 ][interface1][-1, 0, 0] STARTING interface_id=1 2011-05-26 15:27:51.507876 [ERR] gsmopen_protocol.cpp:696 rev [(nil)|37 ][ERRORA 696 ][interface1][-1, 0, 0] no response to AT 2011-05-26 15:27:51.507876 [ERR] mod_gsmopen.cpp:1907 rev [(nil)|37 ][ERRORA 1907 ][interface1][-1, 0, 0] gsmopen_serial_config failed, let's try again 2011-05-26 15:27:55.629412 [ERR] gsmopen_protocol.cpp:696 rev [(nil)|37 ][ERRORA 696 ][interface1][-1, 0, 0] no response to AT 2011-05-26 15:27:55.629412 [ERR] mod_gsmopen.cpp:1913 rev [(nil)|37 ][ERRORA 1913 ][interface1][-1, 0, 0] 1: gsmopen_serial_config failed, let's try again 2011-05-26 15:27:57.854725 [ERR] gsmopen_protocol.cpp:2285 rev [(nil)|37 ][ERRORA 2285 ][interface1][-1, 0, 0] Error RE-sending (carriage return): 10 -1 (Invalid or incomplete multibyte or wide character) 2011-05-26 15:27:57.854725 [ERR] gsmopen_protocol.cpp:2364 rev [(nil)|37 ][ERRORA 2364 ][interface1][-1, 0, 0] Error sending data... (Invalid or incomplete multibyte or wide character) 2011-05-26 15:27:57.854725 [ERR] gsmopen_protocol.cpp:696 rev [(nil)|37 ][ERRORA 696 ][interface1][-1, 0, 0] no response to AT 2011-05-26 15:27:57.854725 [ERR] mod_gsmopen.cpp:1913 rev [(nil)|37 ][ERRORA 1913 ][interface1][-1, 0, 0] 2: gsmopen_serial_config failed, let's try again 2011-05-26 15:28:00.621621 [ERR] gsmopen_protocol.cpp:2259 rev [(nil)|37 ][ERRORA 2259 ][interface1][-1, 0, 0] Error RE-sending (A): 10 -1 (Invalid or incomplete multibyte or wide character) 2011-05-26 15:28:00.621621 [ERR] gsmopen_protocol.cpp:2364 rev [(nil)|37 ][ERRORA 2364 ][interface1][-1, 0, 0] Error sending data... (Invalid or incomplete multibyte or wide character) 2011-05-26 15:28:00.621621 [ERR] gsmopen_protocol.cpp:696 rev [(nil)|37 ][ERRORA 696 ][interface1][-1, 0, 0] no response to AT 2011-05-26 15:28:00.621621 [ERR] mod_gsmopen.cpp:1913 rev [(nil)|37 ][ERRORA 1913 ][interface1][-1, 0, 0] 3: gsmopen_serial_config failed, let's try again ........................ freeswitch(a)localhost.localdomain> 2011-05-26 15:28:05.063925 [ERR] mod_gsmopen.cpp:1917 rev [(nil)|37 ][ERRORA 1917 ][interface1][-1, 0, 0] STARTING interface_id=1 FAILED 2011-05-26 15:28:05.063925 [ERR] mod_gsmopen.cpp:3146 rev [(nil)|37 ][ERRORA 3146 ][interface1][-1, 0, 0] ALARM on interface interface1: 2011-05-26 15:28:05.063925 [CONSOLE] switch_loadable_module.c:946 Successfully Loaded [mod_gsmopen] 2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:145 Adding Endpoint 'gsmopen' 2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:274 Adding API Function 'gsm' 2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:274 Adding API Function 'gsmopen' 2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:274 Adding API Function 'gsmopen_boost_audio' 2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:274 Adding API Function 'gsmopen_dump' 2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:274 Adding API Function 'gsmopen_sendsms' 2011-05-26 15:28:05.063925 [NOTICE] switch_loadable_module.c:379 Adding Chat interface 'SMS' Thank you Tang du -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-load-mod-gsmopen-tp6406… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

Making an outbound call to a cell phone and letting the callee interact with a JavaScript program
by Yungwei Chen 26 May '11

26 May '11

Hi, I am wondering which application allows me to do the following: 1. make an outbound call from a SIP phone to a cell phone. 2. When the call is answered, run a JavaScript application (so the caller can interact with the JavaScript program.) Thanks.

1 0

rtpmap line missing on answer
by Sean Eichhorn 25 May '11

25 May '11

I have a freeswitch system that (for the most part) works flawlessly for me. However, I need to have an additional rtpmap line in the SDP when a call is answered. My client sends the following SDP upon answering the call : m=audio 17214 RTP/AVP 0 19 101 100 c=IN IP4 192.168.98.79 a=rtpmap:0 PCMU/8000 a=rtpmap:19 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 Freeswitch forwards the following SDP to the external gateway : m=audio 20654 RTP/AVP 0 19 101 c=IN IP4 66.223.187.208 a=rtpmap:0 PCMU/8000 a=rtpmap:19 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 The other information appears fine. Everything works, but my client-side is not receiving the a=rtpmap:100 line. This issue appears regardless of whether or not I'm in proxy mode, bypass mode, or neither. Using "sip_append_audio_sdp" has no effect on the answering SDP, only the initial offer. Any ideas? Thanks in advance,

5 10

Playing a wav file on a remote web server
by Yungwei Chen 25 May '11

25 May '11

Hi, I am currently using the following javascript snippet to play a wav file on a remote web server. I'm wondering if there's another way to do it without involving a local file, just like mod_shout. Thanks. function on_dtmf(session, type, digits, arg) { dtmf_digits += digits.digit; console_log("dtmf_digits=" + dtmf_digits + "\n"); return(true); } var ttsUrl = "http://127.0.0.1/tts/getWav.aspx?text=please%20enter%20your%204%20digit%20p…"; var file = "/tmp/tts.wav"; fetchURLFile(ttsUrl,file); session.streamFile(file, on_dtmf);

3 5

DTMF Problems! In-band detection patchy, rfc2833 not working!
by Sidharth Kshatriya 25 May '11

25 May '11

Dear Friends, I'm using a voip carrier Voicenetwork.ca (recommended by someone on this list). While I'm generally happy with their service there seems to be one fatal problem: My IVR does not recognize DTMF! I have set <param name="dtmf-type" value="rfc2833"/> in both sip_profile/internal.xml and sip_profile/external.xml The symptom of the problem is that making a call via skype will *always*make the IVR recognize the DTMF while using something like a mobile phone *almost always* won't! I've tried in-band detection too. I'm making international calls into my IVR and the reliability of the in-band detection is not so good, possibly because of the quality of the call. Can someone please help me? Thanks, Sidharth -- Sidharth Kshatriya www.sidk.info

7 15

FS not recognizing 200 OK after far-end call is answered
by Blake Sevier 25 May '11

25 May '11

Hi all, I’m new, so please bear with me. My setup is fairly basic, and the routing all seems to work just fine: FS receives call from an OpenSIPS proxy, it answers the call, records it and plays back a short prompt before invoking the “bridge” function to place an outbound call via the same OpenSIPS proxy to an external SIP endpoint (in testing, just an Asterisk instance that answers the call). This all works just fine, except FS doesn’t seem to recognize the 200 OK it receives back for leg b after the Asterisk instances answers the call. It apparently does recognize/accept the 100 Trying, as FS doesn’t send additional INVITEs. I’m not seeing any issues with the SIP messaging received by FS – it’s apparently formatted correctly. To note, I’ve also tested calls direct to the Asterisk instance (bypassing OpenSIPS proxy), and get similar results. My guess is that I’ve got something incorrectly configured within my dialplan?? However, not sure how to debug any further to figure out why the 200 Oks are not being recognized. Eventually, after 60s, FS ends the call with a NO_ANSWER. Below is a link to a full call trace from FS – from start of call to when I see multiple 200 OKs from Asterisk. http://pastebin.com/bmEtiW9X Any help would be greatly appreciated. If any further details are needed, please let me know Thank you!!! Blake

1 1

transfer to extension without pattern matching
by Adam Kelloway 25 May '11

25 May '11

Hi there, The wiki page describing transfer (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer) states the following: "Immediately transfer the calling channel to a new context. If there happens to be an xml extension named <destination_number> then control is "warped" directly to that extension. Otherwise it goes through the entire context checking for a match." Is this a correct statement? It seems that even when I try to transfer control directly to an extension, the entire context is still checked for a match. I am using the following: .. <application="transfer" data="test XML default"/> .. <extension name="test"> <condition field="destination_number" expression="^test$"> ... </condition> </extension> Is there a way to disable the pattern matching/dial plan processing? Thanks, Adam

3 3

FreeSWITCH Conference Call Today
by Michael Collins 25 May '11

25 May '11

Hi folks! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_05_25 We have a few news items to discuss and if there's time I will share with you the technique I demonstrated for using bind_digit_action in conferences. Talk to you soon, Michael

1 0

Errors while sending faxes
by Mathieu Lautram 25 May '11

25 May '11

Hi all, After sending faxes (it works very well know, thanks to Anthony Minessale) , sometimes I've got errors like that: - Disconnected after permitted retries - No response after sending a page - Received a DCN from remote after sending a page - Received bad response to DCS ot TCF - The call dropped prematurely - Timed out waiting for initial communication Those faxes are about 25% of all faxes that I send. I'm using the last git version. Where this problem comes from? Is there a way to fix this? I know that sometimes the remote end could be broken or could be an old fax machine, but it seems strange that I have so many errors like that. Thank you for your help :-) -- Mathieu LAUTRAM Application developer BJT Partners - FRANCE +33 1 79 75 99 60 +33 6 61 59 07 25

1 0

Outbound SIP invites without codec list
by Liam Farr 25 May '11

25 May '11

Hi, My outbound SIP invites are going out without a codec list? Could someone point me in the right direction on how to set / insert this? send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565: ------------------------------------------------------------------------ INVITE sip:0800000000@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr Max-Forwards: 69 From: "Liam - Test" <sip:12345678@9.8.7.6>;tag=ryFQea59e6Upe To: <sip:0800000000@1.2.3.4:5060> Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43 CSeq: 12805602 INVITE Contact: <sip:12345678@9.8.7.6:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported 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: 189 X-FS-Support: update_display Remote-Party-ID: "Liam - Test" <sip:12345678@9.8.7.6 >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6 s=FreeSWITCH c=IN IP4 9.8.7.6 t=0 0 m=audio 18646 RTP/AVP 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ Thanks Liam

4 4