December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Call sip phones from gtalk / jabber
by kriko 17 Dec '08

17 Dec '08

I have an idea which is takes too many characters for irc. I'm relatively new to telephony and such stuff, I managed to get freeswitch running, but I don't fully uderstand my problem in detail and how to solve it, so I need a bit of directions. Briefly, my idea is to have a jabber contact, which would gtalk user add as their buddy. Let's call it callbuddy(a)somejabbersrv.com. This would be a bot that would accept commands - I've already made a small xmpp java bot which just logs into gtalk and send echo messages to users - like a parrot, nothing serious. Also I've set up openfire jabber server. What I would like him to do, is when user would tipe "call 0189432443" it would initiate a call between contact who tiped in this number (command) and this number. Is this possible, and what would be the best implementation (dingaling acting as client...)? Thanks. -- kriko

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general question about API command
by Carole O. 17 Dec '08

17 Dec '08

Hello, I have a general question about the API commands. Some of them are not available in the dialplan like uuid_transfer. I was wondering how to call an API command without using the CLI. Especially I would be interested in knowing if there is any way to call them from a phone, I mean bind a key to an API command. For instance I would like to transfer both members of a simple call into a conference by dialing *1. I have seen the transfer is possible from the CLI by doing: api uuid_transfer <uuid> -both 3001 but I do not know how to do it else. If somebody could give me an insight about the topic it would be great. Thanks, Carole -- View this message in context: http://www.nabble.com/general-question-about-API-command-tp21032754p2103275… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Speed Dial Emulation
by Michael S Collins 16 Dec '08

16 Dec '08

Do you need something just for one extension? Or system wide? If it's system wide then all you need is an extension that matches a condition like this: <condition field="destination_number" expression="^1$"> Anyone who dials just a single digit 1 will go to this Dialplan entry. From there you can have it bridge to whatever endpoint you like. -MC Sent from my iPhone On Dec 15, 2008, at 7:45 AM, jflowers <jflowers(a)ezo.net> wrote: > > How do I emulate a speed dial setup. That is, from extension 1003 I > dial > just a 1 ( or 2, or 3 etc.) and nothing else and freeswitch dials a > PSTN > number. Is there software to do this? > > > -- > View this message in context: http://www.nabble.com/Speed-Dial-Emulation-tp21016167p21016167.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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FreeSWITCH Infrastructure / Happy Holidays!
by Brian West 16 Dec '08

16 Dec '08

FreeSWITCers, As the project grows I felt the need for the project to own the critical infrastructure hosting things like SVN, Jira, Fisheye and various other things we as a project use. I have personally paid for two servers that Bandwidth.com is going to be colocating for the project in their data center. Teliax paid for and shipped me the two 3ware raid1 cards for the machines which was a great help. The servers are going into production after the first of the year and in the process we are rebuilding everything to allow for single sign on for every service on the FreeSWITCH project. The servers are in the hands of bandwidth.com to be racked up so we can start deployment, as you can see we have our work cut out for us with the integration and rebuild of our entire infrastructure. ;) If you wish to pitch in my personal paypal is brian(a)freeswitch.org or you can use chipin http://www.chipin.com/contribute/id/ddc094318ae5ab5b I would like to thank Bandwidth.com and Teliax for their support... I hope others will help out! Also we have another sounds order going in sometime near the end of January so any money that exceeds the server cost will go to that. The sound order I placed monday will be in this Friday. Happy Holidays and wish you the best for 2009! Thanks, Brian West FreeSWITCH

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Ending a bridged call with a touchtone
by Marc Orenberg 16 Dec '08

16 Dec '08

Hello. I'm trying to allow the A-leg of a bridged call to be able to press a touchtone to end the call. In my Python script, I set-up a DTMF callback function using setInputCallback, but it doesn't seem to have any effect during bridged calls. Is there another way to do this? Thanks.

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FS mod_fax
by Helmut Kuper 16 Dec '08

16 Dec '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried to compile mod_fax today with trunk from yesterday. A 'make' in FS trunk directory led to an error saying that libspandsp.la wasn't found in libs/spandsp/src. So I had to configure and compile (make) spandsp manually before compiling FS. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAkk+Q18ACgkQ4tZeNddg3dw5GgCgmmuLCsAx+T7IzUPayqAXZDaa BO8AoLa5wUOBqaEG1pOG4Qow8r7J2NF7 =MOUJ -----END PGP SIGNATURE-----

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OpenZAP: "Received unhandled message 125"
by Peter P GMX 16 Dec '08

16 Dec '08

I have installed OpenZAP with a TE220 card and EuroISDN. When I bridge an outgoing call I get a "Received unhandled message 125" Any idea what that means? As far as I know there is no <external> result code 125 defined in the ISDN protocol. Best regards Peter See the following logs. 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 14 -------------------------------------------------------------------------------- [08 02 80 03 7d 08 04 82 e3 98 28 14 01 01] 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Received message from Q.921 (ind 4, tei 0, size 18) MesType: 125, CRVFlag 1 (Terminator), CRV 3 (Dialect: 2) 2008-11-27 17:35:17 [DEBUG] Span:0 Q.931() Sending message to Layer4 (size: 114) 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[7d] Size:[114] CRV: 3 (0x3, CTX: Terminator) 2008-11-27 17:35:17 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 70d230 (1:1) source isdn_data->channels_local_crv[0x3] 2008-11-27 17:35:17 [CRIT] ozmod_isdn.c:760 zap_isdn_931_34() Received unhandled message 125 (0x7d) 2008-11-27 17:35:17 [DEBUG] Span:0 Q.921() Sending frame >oz list API CALL [oz(list)] output: +OK span: 1 (span1) type: isdn chan_count: 31 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none +OK span: 2 (span2) type: isdn chan_count: 31 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none

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mod_ldap
by Helmut Kuper 16 Dec '08

16 Dec '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I updated to latest FS trunk today and tried to compile mod_ldap and got this errors: mkdir .libs ar cru .libs/libldap.a bind.o open.o result.o error.o compare.o search.o controls.o messages.o references.o extended.o cyrus.o modify.o add.o modrdn.o delete.o abandon.o sasl.o sbind.o kbind.o unbind.o cancel.o filter.o free.o sort.o passwd.o whoami.o getdn.o getentry.o getattr.o getvalues.o addentry.o request.o os-ip.o url.o sortctrl.o vlvctrl.o init.o options.o print.o string.o util-int.o schema.o charray.o tls.o os-local.o dnssrv.o utf-8.o utf-8-conv.o turn.o groupings.o txn.o ppolicy.o version.o ranlib .libs/libldap.a creating libldap.la (cd .libs && rm -f libldap.la && ln -s ../libldap.la libldap.la) cc -g -O2 -o apitest apitest.o ./.libs/libldap.a /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/liblber/.libs/liblber.a ../../libraries/liblber/.libs/liblber.a ../../libraries/liblutil/liblutil.a /usr/lib/libsasl2.a -ldl -lresolv ./.libs/libldap.a(os-ip.o): In function `ldap_pvt_is_socket_ready': /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194: warning: `sys_errlist' is deprecated; use `strerror' or `strerror_r' instead /opt/app/voip/src/freeswitch/trunk/libs/openldap-2.3.19/libraries/libldap/os-ip.c:194: warning: `sys_nerr' is deprecated; use `strerror' or `strerror_r' instead /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_close': (.text+0x348): undefined reference to `db_strerror' /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open': (.text+0x3d0): undefined reference to `db_create' /usr/lib/libsasl2.a(db_berkeley.o): In function `berkeleydb_open': (.text+0x453): undefined reference to `db_strerror' /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_getdata': (.text+0x7e8): undefined reference to `db_strerror' /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata': (.text+0xa4e): undefined reference to `db_strerror' /usr/lib/libsasl2.a(db_berkeley.o): In function `_sasldb_putdata': (.text+0xafa): undefined reference to `db_strerror' /usr/lib/libsasl2.a(digestmd5.o): In function `init_des': (.text+0x6c8): undefined reference to `DES_key_sched' /usr/lib/libsasl2.a(digestmd5.o): In function `init_des': When I change src/mod/directories/mod_ldap/Makefile line 3 to "LDAP=openldap-2.4.11" and putting openldap-2.4.11.tgz in libs/ manually it compiles without error, but with some warnings. Maybe the openldap-2.3.19 delivered by FS is too old? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklHp/MACgkQ4tZeNddg3dzjOgCfSExGU7PSWSdAxI4VCZIosrLJ 25kAoK0igc53iV1PhWkj5faAVOL78E8B =ny1E -----END PGP SIGNATURE-----

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Openzap ERROR can't dial
by fidibus83 16 Dec '08

16 Dec '08

Have nobody an idea?

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Bind error
by Jonas Gauffin 16 Dec '08

16 Dec '08

I get a bind error for the RTP, can someone be kind and explain why? 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:2388 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000]/[PCMA:8:8000] 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1596 sofia_glue_tech_set_codec() Set Codec sofia/internal/0236661201(a)sip-corporate2.tele2.se PCMA/8000 20 ms 160 samples 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:2352 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1814 sofia_glue_activate_rtp() AUDIO RTP [sofia/internal/0236661201(a)sip-corporate2.tele2.se] 192.168.1.112 port 17102 -> 130.244.5X.XX port 27354 codec: 8 ms: 20 2008-12-16 10:47:07 [DEBUG] switch_rtp.c:858 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media() Ring-Ready sofia/internal/0236661201(a)sip-corporate2.tele2.se! 2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2097 sofia_glue_tech_media() Pre-Answer sofia/internal/0236661201(a)sip-corporate2.tele2.se! 2008-12-16 10:47:07 [DEBUG] switch_channel.c:1615 switch_channel_perform_pre_answer() sofia/external/0707992871(a)212.151.14X.XXX receive message [SWITCH_MESSAGE_INDICATE_PROGRESS] 2008-12-16 10:47:07 [INFO] mod_sofia.c:1253 sofia_receive_message() Asked to send early media by sofia/external/0707992871(a)212.151.14X.XXX 2008-12-16 10:47:07 [DEBUG] switch_channel.c:1585 switch_channel_perform_mark_pre_answered() Send signal sofia/external/0707992871(a)212.151.14X.XXX [BREAK] 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:497 sofia_glue_ext_address_lookup() STUN Success [85.89.7X.XXX]:[33594] 2008-12-16 10:47:07 [DEBUG] sofia_glue.c:1814 sofia_glue_activate_rtp() AUDIO RTP [sofia/external/0707992871(a)212.151.14X.XXX] 192.168.1.112 port 17108 -> 130.244.1X.XXX port 17148 codec: 8 ms: 20 2008-12-16 10:47:07 [DEBUG] switch_rtp.c:858 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2008-12-16 10:47:07 [ERR] sofia_glue.c:2045 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-16 10:47:07 [NOTICE] sofia_glue.c:2046 sofia_glue_activate_rtp() Hangup sofia/external/0707992871(a)212.151.14X.XXX [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-12-16 10:47:07 [DEBUG] switch_channel.c:1478 switch_channel_perform_hangup() Send signal sofia/external/0707992871(a)212.151.14X.XXX [KILL] 2008-12-16 10:47:07 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/external/0707992871(a)212.151.14X.XXX [BREAK] 2008-12-16 10:47:07 [INFO] mod_sofia.c:1294 sofia_receive_message() Ring SDP: v=0 o=FreeSWITCH 1229387233 1229387234 IN IP4 85.89.7X.XXX s=FreeSWITCH c=IN IP4 85.89.7X.XXX t=0 0 m=audio 33594 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2008-12-16 10:47:07 [NOTICE] mod_sofia.c:1297 sofia_receive_message() Ring-Ready sofia/external/0707992871(a)212.151.14X.XXX! 2008-12-16 10:47:07 [NOTICE] mod_sofia.c:1297 sofia_receive_message() Pre-Answer sofia/external/0707992871(a)212.151.14X.XXX! 2008-12-16 10:47:07 [DEBUG] switch_core_session.c:510 switch_core_session_perform_receive_message() Send signal sofia/external/0707992871(a)212.151.14X.XXX [BREAK] 2008-12-16 10:47:07 [DEBUG] switch_ivr_originate.c:1625 switch_ivr_originate() Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2008-12-16 10:47:07 [NOTICE] switch_ivr_originate.c:1666 switch_ivr_originate() Hangup sofia/internal/ 0236661201(a)sip-corporate2.tele2.se [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2008-12-16 10:47:07 [DEBUG] switch_channel.c:1478 switch_channel_perform_hangup() Send signal sofia/internal/ 0236661201(a)sip-corporate2.tele2.se [KILL] 2008-12-16 10:47:07 [DEBUG] switch_core_session.c:806 switch_core_session_signal_state_change() Send signal sofia/internal/ 0236661201(a)sip-corporate2.tele2.se [BREAK] 2008-12-16 10:47:07 [INFO] mod_dptools.c:1869 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL

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