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Issue with playAndGetDigit
by Juan Backson 12 Nov '08

12 Nov '08

Hi, I am encountering an issue when using playAndGetDigit. I have the regex set as 1|2|3|4|5|6, and then pressed 9, it gives error as expected. Then, I pressed 2. This time it passes, but the digit being returned by playAndGetDigit is 92, instead of just 2. Also, the voice prompt seems a bit strange as well. For the first time I entered an invalid response, I would hear just the invalid prompt defined as one of the parameter, but for the second time I entered an invalid response, I would hear both the invalid prompt and the normal prompt. Is this normal? JB

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FACILITY_NOT_SUBSCRIBED
by Peter P GMX 12 Nov '08

12 Nov '08

I receive a FACILITY_NOT_SUBSCRIBED message when I call from 1000 to 1001. I've read in the wiki: 50 FACILITY_NOT_SUBSCRIBED requested facility not subscribed [Q.850 This cause indicates that the user has requested a supplementary service, which is available, but the user is not authorized to use. I am wondering which supplementary service this could be. The invite message is as follows: ====================== U xxx.xx.xx.186:2054 -> xxx.xx.xxx.xx:5060 INVITE sip:1001@xxx.xx.xxx.xx;user=phone SIP/2.0. Via: SIP/2.0/UDP xxx.xx.xx.186:2054;branch=z9hG4bK-7qnms1c9aoqt;rport. From: "Company1000" <sip:1000@xxx.xx.xxx.xx>;tag=pslbxvhxjo. To: <sip:1001@xxx.xx.xxx.xx;user=phone>. Call-ID: 3c267cac8c1f-smz6hpvv1e5h. CSeq: 2 INVITE. Max-Forwards: 70. Contact: <sip:1000@xxx.xx.xx.186:2054;line=rt3bxln1>;flow-id=1. P-Key-Flags: keys="3". User-Agent: snom320/7.1.33. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer, call-info. Supported: timer, 100rel, replaces, from-change. Session-Expires: 3600;refresher=uas. Min-SE: 90. Proxy-Authorization: Digest username="1000",realm="xxx.xx.xxx.xx",nonce="201af2a0-af3f-11dd-af96-ebbe56552456",uri="sip:1001@xxx.xx.xxx.xx;user=phone",qop=auth,nc=00000001,cnonce="0f67abbc",response="d984661b55085f64e3e6f93a09762eca",algorithm=MD5. Content-Type: application/sdp. Content-Length: 370. . v=0. o=root 1618416056 1618416056 IN IP4 xxx.xx.xx.186. s=call. c=IN IP4 xxx.xx.xx.186. t=0 0. m=audio 12472 RTP/AVP 8 0 9 2 3 18 4 101. a=rtpmap:8 pcma/8000. a=rtpmap:0 pcmu/8000. a=rtpmap:9 g722/8000. a=rtpmap:2 g726-32/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 g729/8000. a=rtpmap:4 g723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. ====================== So there is nothing special in the SDP. Both Snom 320 phones register on the right domain (here:IP). They both can call their mailbox etc. They are behind a NAT on different public IPs. However calling each other doesn't work. Sofia status profile internal API CALL [sofia(status profile internal)] output: ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP xxx.xx.xxx.xx SIP-IP xxx.xx.xxx.xx URL sip:mod_sofia@xxx.xx.xxx.xx:5060 BIND-URL sip:mod_sofia@xxx.xx.xxx.xx:5060 HOLD-MUSIC local_stream://moh CODECS G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN_ENABLED true STUN_AUTO_DISABLE false Registrations: ================================================================================================= Call-ID: 3c2679b578bd-8brbg608itvr User: 1000(a)xxx.xx.xxx.xx Contact: "Company1000" <sip:1000@xxx.xx.xx.186:2054;line=145ehzt5> Agent: snom320/7.1.33 Status: Registered(UDP)(unknown) EXP(2008-11-10 16:45:40) Host: freeswitch Call-ID: 3c2677883993-68zb6go2xoip User: 1001(a)xxx.xx.xxx.xx Contact: "Company1001" <sip:1001@xxx.xx.xx.189:2054;line=hcg076gv> Agent: snom320/7.1.33 Status: Registered(UDP)(unknown) EXP(2008-11-10 16:45:41) Host: freeswitch ================================================================================================= What can I do? Best regards Peter

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Receiving 406 From Freeswitch....Any Clues?
by Adeel Ansari 12 Nov '08

12 Nov '08

Hi, I am getting 406, no matter what. I have tried 3 different INVITEs. ------ *INVITE sip:1002@192.168.253.101 <sip%3A1002(a)192.168.253.101>;transport=udp SIP/2.0 Call-ID: ef1e16976ae32f0f011de0db2ab5804b(a)192.168.253.101 CSeq: 1 INVITE From: <sip:1001@192.168.253.101 <sip%3A1001(a)192.168.253.101>>;tag=5919 To: <sip:1002@192.168.253.101 <sip%3A1002(a)192.168.253.101>> Via: SIP/2.0/UDP 192.168.253.101:7620 ;branch=z9hG4bKcfac6aaa7ee4a96457335bdad787cf31 Max-Forwards: 2 Contact: <sip:1001@192.168.253.101:7620;transport=udp> Accept: audio/gsm,audio/x-gsm,text/plain Content-Length: 0 *------ ** and I tried, ------- *INVITE sip:1002@192.168.253.101 <sip%3A1002(a)192.168.253.101>;transport=udp SIP/2.0 Call-ID: 7a612541e08e583e0069c988c8323425(a)192.168.253.101 CSeq: 1 INVITE From: <sip:1001@192.168.253.101 <sip%3A1001(a)192.168.253.101>>;tag=7387 To: <sip:1002@192.168.253.101 <sip%3A1002(a)192.168.253.101>> Via: SIP/2.0/UDP 192.168.253.101:6453 ;branch=z9hG4bK7ccbc7f8e7bc0c4ac0483cca54a48489 Max-Forwards: 2 Contact: <sip:1001@192.168.253.101:6453;transport=udp> Accept: audio/gsm,audio/x-gsm,text/plain Authorization: Digest username="1001",realm="192.168.253.101",uri="sip: 192.168.253.101:5060 ",algorithm=MD5,opaque="",nonce="b1f5e8d8-afd0-11dd-ad17-f7c4d6a988fd",response="2c70f260de2292d9e01406bdd1f90e28" Content-Length: 0 *------- and then I tried, ------ *INVITE sip:1002@192.168.253.101 <sip%3A1002(a)192.168.253.101>;transport=udp SIP/2.0 Call-ID: 12e7c3af8d94ba048789f75e7036ea74(a)192.168.253.101 CSeq: 1 INVITE From: <sip:1001@192.168.253.101 <sip%3A1001(a)192.168.253.101>>;tag=6309 To: <sip:1002@192.168.253.101 <sip%3A1002(a)192.168.253.101>> Via: SIP/2.0/UDP 192.168.253.101:4487 ;branch=z9hG4bK3282799ef902293b7a4795d1cc3d78bd Max-Forwards: 2 Contact: <sip:1001@192.168.253.101:4487;transport=udp> Accept: audio/gsm,audio/x-gsm,text/plain Authorization: Digest username="1001",realm="192.168.253.101",uri="sip: 192.168.253.101:5060 ",algorithm=MD5,opaque="",nonce="85c12b22-afd2-11dd-ad17-f7c4d6a988fd",response="c8d6083112fc68b6e53d7e5b38e990eb" Content-Type: application/sdp Content-Length: 111 v=0 o=1001 279445 280814 IN IP4 192.168.253.101 s=- c=IN IP4 192.168.253.101 t=0 0 m=audio 1436 RTP/AVP *------ But I always receive the same 406. Any idea. The response I am receiving from Freeswitch is, ------- *received response : SIP/2.0 406 Not Acceptable Via: SIP/2.0/UDP 192.168.253.101:4487 ;branch=z9hG4bK3282799ef902293b7a4795d1cc3d78bd From: <sip:1001@192.168.253.101 <sip%3A1001(a)192.168.253.101>>;tag=6309 To: <sip:1002@192.168.253.101 <sip%3A1002(a)192.168.253.101> >;tag=H1DH6S41c0U2N Call-ID: 12e7c3af8d94ba048789f75e7036ea74(a)192.168.253.101 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9256M Accept: application/sdp Accept-Encoding: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,MESSAGE,SUBSCRIBE,NOTIFY,REFER,UPDATE,REGISTER,INFO,PUBLISH Supported: 100rel,timer,precondition,path,replaces Allow-Events: talk,presence,dialog,call-info,sla,include-session-description,presence.winfo,message-summary Content-Length: 0 *------- Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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Max Number of files sent to session.execute("playback", file)
by fgastiaz 12 Nov '08

12 Nov '08

Hi Guys, I have this problem when i'm trying to reproduce more than 8 prompts in a row. Here is code extract that simplifies the problem i'm having. If put something like this inside a script.py session.execute("playback","1.wav") session.execute("playback","2.wav") session.execute("playback","3.wav") session.execute("playback","4.wav") session.execute("playback","5.wav") session.execute("playback","6.wav") session.execute("playback","7.wav") session.execute("playback","8.wav") session.execute("playback","9.wav") The file 9.wav is not heard even though in the freeswitch.log it says that the file was sent to the module 2008-11-11 18:53:03 [DEBUG] switch_cpp.cpp:570 execute() CoreSession::execute. app: playback data:9.wav 2008-11-11 18:53:03 [DEBUG] switch_ivr_play_say.c:912 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2008-11-11 18:53:03 [DEBUG] switch_core_session.c:430 switch_core_session_receive_message() Kill sofia/internal/221122(a)201.216.201.133 [BREAK] 2008-11-11 18:53:03 [INFO] switch_rtp.c:1253 rtp_common_read() Auto Changing port from 201.216.201.135:8000 to 201.216.201.133:18648 2008-11-11 18:53:03 [DEBUG] switch_ivr_play_say.c:1203 switch_ivr_play_file() done playing file So do you have an idea why after the prompt number 8 the rest are not heard? I will thank any helps on this -- View this message in context: http://www.nabble.com/Max-Number-of-files-sent-to-session.execute%28%22play… Sent from the Freeswitch-users mailing list archive at Nabble.com.

3 3

Problems with ./configure on OSX
by martin joseph 12 Nov '08

12 Nov '08

Hey FreeSwitchers, I have been running 1.01 (which I built from a tarball) without any problems. I see that we are pushing to 1.02, so I tried to do a make current on my trunk source and got the following... /usr/src/freeswitch/trunk root# make current /bin/sh ./config.status --recheck running CONFIG_SHELL=/bin/sh /bin/sh ./configure --no-create --no- recursion checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for a thread-safe mkdir -p... build/config/install-sh -c -d checking for gawk... no checking for mawk... no checking for nawk... no checking for awk... awk checking whether make sets $(MAKE)... yes checking build system type... powerpc-apple-darwin8.11.0 checking host system type... powerpc-apple-darwin8.11.0 checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking for style of include used by make... GNU checking dependency style of gcc... gcc3 checking for g++... g++ checking whether we are using the GNU C++ compiler... yes checking whether g++ accepts -g... yes checking dependency style of g++... gcc3 checking for gawk... (cached) awk checking whether make sets $(MAKE)... (cached) yes checking for a BSD-compatible install... /usr/bin/install -c checking whether gcc and cc understand -c and -o together... yes ./configure: line 4533: syntax error near unexpected token `build_libtool_libs,' ./configure: line 4533: ` _LT_DECL(build_libtool_libs, enable_shared, 0,' make: *** [config.status] Error 2 mail:/usr/src/freeswitch/trunk root# Any ideas on why this might be? Thanks, Marty

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Re: [Freeswitch-users] mod_python sqlalchemy core dumps
by . 12 Nov '08

12 Nov '08

Hi, It seems (still need to run a few more tests to be sure) you have completely solved my problem. This had been driving me nuts for the last several days, and I can't tell you how helpful this simple suggestion was! >I've used Django/psycopg + mod_python successfully in production and it >never crashes. (never tried sqlalchemy). >One thing you could try is to remove the hangup hook. That has been >problematic in the past and not well tested AFAIK. I doubt that's the >problem but you never know. After I removed the hangup hook all of my problems disappeared. Maybe it's worth adding something to the freeswitch mod_python wiki page about this? I still don't know exactly why this causes a core dump, and it seems to work ok, as long as there is nothing else interacting with it (e.g. sqlalchemy), but it probably would have taken me quite a while longer to get to this on my own, given the prominent placing of the hangup_hook in most of the examples on the wiki. >Also please load the core dump in gdb and post a backtrace.. maybe that will >reveal something interesting. (or maybe it won't). Actually you might want >to put it in a jira bug in case it becomes an involved debugging process. I did this as well, after recompiling freeswitch with # export CFLAGS="-g -ggdb" # export MOD_CFLAGS="-g -ggdb" as suggested in the wiki, however the results didnt seem very helpful: (gdb) core core.4051 (no debugging symbols found) Using host libthread_db library "/lib/tls/i686/cmov/libthread_db.so.1". Core was generated by `/usr/local/freeswitch/bin/freeswitch'. Program terminated with signal 11, Segmentation fault. #0 0x00e140df in ?? () >> I've spent hours hunting through related lists: mod_python (for >> apache), sqlalchemy, etc. but all to no avail. Any suggestions or >> alternatives ( I've also started looking into outbound event socket, >> but there still seems to be few people using this with python/twisted >> ) will be greatly appreciated. >Yeah an outbound event socket framework in python would be really cool. >deally you could write a script in the same style as mod_python scripts, >and the framework behind it would handle all the communication with the >switch. Kamealia might be a bettter networking library than Twisted for >this one. I still think this could be very nice, and useful. I'm not familiar with kamealia at all. if you don't mind, could you elaborate on why you think it would be preferable to twisted in this case? Thanks again!

1 0

Re: [Freeswitch-users] mod_python sqlalchemy core dumps ( .)
by Traun Leyden 12 Nov '08

12 Nov '08

I've used Django/psycopg + mod_python successfully in production and it never crashes. (never tried sqlalchemy). One thing you could try is to remove the hangup hook. That has been problematic in the past and not well tested AFAIK. I doubt that's the problem but you never know. Also please load the core dump in gdb and post a backtrace.. maybe that will reveal something interesting. (or maybe it won't). Actually you might want to put it in a jira bug in case it becomes an involved debugging process. > > I've spent hours hunting through related lists: mod_python (for > apache), sqlalchemy, etc. but all to no avail. Any suggestions or > alternatives ( I've also started looking into outbound event socket, > but there still seems to be few people using this with python/twisted > ) will be greatly appreciated. Yeah an outbound event socket framework in python would be really cool. Ideally you could write a script in the same style as mod_python scripts, and the framework behind it would handle all the communication with the switch. Kamealia might be a bettter networking library than Twisted for this one. HTH

1 0

mod_python sqlalchemy core dumps
by . 12 Nov '08

12 Nov '08

Hi, I've been having some frustrations with mod_python lately. I've stripped down the script I've been using to the barest of bare bones, and am still getting core dumps; so I've decided to share it in the hopes that someone else will be able to see where I've gone wrong. The stripped down script below does nothing more than answer, lookup a caller based on their caller_id_number and then play back some audio. I'm pretty certain that the issue is with sqlalchemy, but I'm still not clear as to why sqlalchemy should cause freeswitch to dump core and die. The database model I'm using with sqlalchemy is also quite simple, and has worked fine for almost a year with a corresponding pylons application, so I'm fairly confident that the model itself is not the underlying problem. My best guess thus far is that sqlalchemy is somehow tripping up mod_python and then freeswitch with the way it attempts to maintain connections/state across calls? In any case my setup is as follows: Python Version: 2.4.4 sqlalchemy v0.5.0rc2 freeswitch (1.0.1) [from changelog, compiled from src] Debian 4.0 etch The script I'm running (trying to run right now): --------------------------------------------------------------------------------------- # -*- mode: python; coding: utf-8 -*- import os, re, sys from freeswitch import * from datetime import datetime from time import strftime, gmtime from loadmodel import model consoleLog("info", "Current Python Version: %s\n\n" % str(sys.version)) def hangup_hook( session, what, caller_id ): """ Default hangup hook. Overload as needed. """ consoleLog("info","The application hung up for %s.\n\n" % what) return def input_callback(session, what, obj, arg): """ Default input callback for dtmf and other events. Overload this as needed. """ if (what == "dtmf"): consoleLog("info", what + " " + obj.digit + "\n") try: return arg[obj.digit] except: return None # will make the streamfile audio stop else: consoleLog("info", what + " " + obj.serialize() + "\n") return "pause" def get_current_caller( session ): """ Dig up the current caller in the database. if the caller isn't registered return null. """ current_caller = model.Session.query(model.Caller).filter_by(phone=session.getVariable('caller_id_number')).first() if current_caller==None: current_caller = model.Session.query(model.Caller).filter_by(phone="123456789").first() return current_caller class dummy_caller: def __init__(self, user_id="0", phone="123456789"): self.user_id = user_id self.phone = phone def handler( session, args ): session.answer( ) playback_dtmf = {"1":"pause","3":"stop","7":"seek:-500","9":"seek:+500"} recording_dtmf = {"2":"stop","3":"stop","5":"stop","6":"stop"} guidance_dtmf = {"3":"stop","4":"stop","9":"stop"} #3 upload, 4 replay, 9 re-record session.setVariable( "intro", "/usr/local/freeswitch/sounds/ja/test/test_intro_padded.wav" ) session.setVariable( "guidance", "/usr/local/freeswitch/sounds/ja/test/test_annai_padded.wav" ) session.setVariable( "goodbye", "/usr/local/freeswitch/sounds/ja/test/test_thanks_padded.wav" ) session.setVariable( "beep", "/usr/local/freeswitch/sounds/beeps/beep_fm.wav" ) current_caller = get_current_caller( session ) session.setHangupHook( hangup_hook, str(current_caller.user_id) ) session.streamFile( session.getVariable("intro") ) #make_recording( session, recording_dtmf, session.getVariable("intro"), process=True ) session.streamFile( session.getVariable("goodbye") ) session.hangup( ) --------------------------------------------------------------------------------------- After 2 or sometimes 3 calls this inevitably causes the following crash: --------------------------------------------------------------------------------------- 2008-11-12 12:40:08 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/external/+818053753079(a)voxbone.com [9a56541a-b06b-11dd-8a29-191ef9635961] 2008-11-12 12:40:08 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing +XXXXXXXX->9993 in context default Segmentation fault (core dumped) --------------------------------------------------------------------------------------- I've spent hours hunting through related lists: mod_python (for apache), sqlalchemy, etc. but all to no avail. Any suggestions or alternatives ( I've also started looking into outbound event socket, but there still seems to be few people using this with python/twisted ) will be greatly appreciated. If all else fails I can probably switch to perl or js but I'd much prefer to stick to python if there is a workable solution ( or -hopefully- if I've just missed something very obvious ). Cheers

1 0

getDigits always missing the first digit
by Cristian Talle 12 Nov '08

12 Nov '08

Hi, I'm having some issues retrieving a dtmf sequence using getDigits. The xml dialplan invokes a javascript file that in turn runs: var digits = ""; session.answer(); digits = session.getDigits(6,"#", 5000); console_log("info", "caller entered: " + digits); ... session.hangup(); The digits variable is always missing the first digit, no matter what I do. I've added some more debug messages, it looks like switch_channel_queue_dtmf() is only called from the second digit on. Has anyone experienced this before? Thanks, Cristian

2 1

[freeswitch-users] Is anyone familiar with the proxy package - Liverpie
by jonathan augenstine 11 Nov '08

11 Nov '08

I would like to know if anyone is familiar with the Freeswitch proxy package, Liverpie(http://www.liverpie.com)? If so, what have your experiences been with this package? Jonathan

1 0