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strange errors when using mod_python
by hkalyoncu 14 Mar '11

14 Mar '11

hello, im doing dynamic routing with mod_python. my python script returns a dialplan after checking db record in terms of destination number. the problem is: although python script returns correct result, freeswitch cannot process returned dialplan and gives strange errors. This is not happening in all calls but i can say 3-4 out of 10 calls drop like this. i rebuilt freeswitch with latest repository but there is no change. here is a sample output from console(sorry for censored fields): 2011-03-11 14:20:49.073690 [INFO] mod_dialplan_xml.c:331 Processing xxxxxxxxxxxx ->xxxxxxxxxxxx in context context_3 2011-03-11 14:20:49.074747 [NOTICE] mod_python.c:118 Invoking py module: dp 2011-03-11 14:20:49.074747 [DEBUG] mod_python.c:188 Call python script 2011-03-11 14:20:49.074747 [INFO] switch_cpp.cpp:1197 calling number : xxxxxxxxxx 2011-03-11 14:20:49.074747 [INFO] switch_cpp.cpp:1197 called number : xxxxxxxxxx 2011-03-11 14:20:49.075833 [DEBUG] mod_python.c:191 Finished calling python script Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195 parsing [context_3->generated] continue=false Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195 Absolute Condition [generated] Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195 Action answer() Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195 Action bridge(sofia/sipinterface_5/xxxxxxxxxx@10.10.1.5:5060) Dialplan: sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195 Action () 2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:119 (sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195) State Change CS_ROUTING -> CS_EXECUTE 2011-03-11 14:20:49.075833 [DEBUG] switch_core_session.c:1116 Send signal sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195 [BREAK] 2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:359 (sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195) State ROUTING going to sleep 2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:320 (sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195) Running State Change CS_EXECUTE 2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:366 (sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195) State EXECUTE 2011-03-11 14:20:49.075833 [DEBUG] mod_sofia.c:240 sofia/sipinterface_4/xxxxxxxxxxxxx(a)xxx.22.9.195 SOFIA EXECUTE 2011-03-11 14:20:49.075833 [DEBUG] switch_core_state_machine.c:157 sofia/sipinterface_4/xxxxxxxxxx(a)xxx.22.9.195 Standard EXECUTE EXECUTE sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195 answer() 2011-03-11 14:20:49.075833 [DEBUG] sofia_glue.c:2990 AUDIO RTP [sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195] 10.xxx.xxx.xxx port 31882 -> xxx.xxx.xxx.xxx port 35892 codec: 18 ms: 20 2011-03-11 14:20:49.075833 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms 2011-03-11 14:20:49.076915 [DEBUG] sofia_glue.c:3245 Set 2833 dtmf send payload to 101 2011-03-11 14:20:49.076915 [DEBUG] sofia_glue.c:3250 Set 2833 dtmf receive payload to 101 2011-03-11 14:20:49.076915 [DEBUG] mod_sofia.c:681 Local SDP sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195: v=0 o=FreeSWITCH 1299814167 1299814168 IN IP4 10.xxx.xxx.xxx s=FreeSWITCH c=IN IP4 10.xxx.xxx.xxx t=0 0 m=audio 31882 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2011-03-11 14:20:49.076915 [DEBUG] switch_core_session.c:709 Send signal sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195 [BREAK] 2011-03-11 14:20:49.076915 [DEBUG] switch_channel.c:2796 (sofia/sipinterface_4/xxxxxxxxxxxxx(a)xxx.22.9.195) Callstate Change RINGING -> ACTIVE 2011-03-11 14:20:49.077991 [NOTICE] mod_dptools.c:929 Channel [sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195] has been answered 2011-03-11 14:20:49.077991 [DEBUG] sofia.c:4725 Channel sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195 entering state [completed][200] 2011-03-11 14:20:49.077991 [ERR] switch_core_session.c:1918 Invalid Application .22.9.195 Action bridge(sofia/sipinterface_5/xxxxxxxxxxxx@10.10.1.5:5060) 2011-03-11 14:20:49.077991 [DEBUG] switch_channel.c:2546 (sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195) Callstate Change ACTIVE -> HANGUP 2011-03-11 14:20:49.077991 [NOTICE] switch_core_session.c:1919 Hangup sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2011-03-11 14:20:49.077991 [DEBUG] switch_channel.c:2562 Send signal sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195 [KILL] 2011-03-11 14:20:49.077991 [DEBUG] switch_core_session.c:1116 Send signal sofia/sipinterface_4/xxxxxxxxxxx(a)xxx.22.9.195 [BREAK] 2011-03-11 14:20:49.077991 [DEBUG] switch_core_state_machine.c:366 (sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195) State EXECUTE going to sleep 2011-03-11 14:20:49.077991 [DEBUG] switch_core_state_machine.c:320 (sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195) Running State Change CS_HANGUP 2011-03-11 14:20:49.077991 [DEBUG] switch_core_state_machine.c:560 (sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195) State HANGUP 2011-03-11 14:20:49.077991 [DEBUG] mod_sofia.c:457 Channel sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195 hanging up, cause: DESTINATION_OUT_OF_ORDER 2011-03-11 14:20:49.077991 [DEBUG] mod_sofia.c:500 Sending BYE to sofia/sipinterface_4/xxxxxxxxxxxx(a)xxx.22.9.195 The text in error message (Invalid Application xxxxx) changes in every call. i recheck the python script several times, each time it gives expected xml output. But freeswitch somehow cannot get (or process?) that output. Any idea? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/strange-errors-when-using-mod… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 6

Compile error on Suse 11.3
by Sascha Daniels 14 Mar '11

14 Mar '11

Hi together. I did build FS on different systems without any problems. On Suse 11.3 (32bit) I get the following error: Compiling src/switch_core_sqldb.c ... cc1: warnings being treated as errors src/switch_core_sqldb.c: In function '_switch_cache_db_get_db_handle': src/switch_core_sqldb.c:314:47: error: comparison between 'switch_odbc_status_t' and 'enum <anonymous>' src/switch_core_sqldb.c: In function 'switch_cache_db_execute_sql_real': src/switch_core_sqldb.c:389:90: error: comparison between 'switch_status_t' and 'enum <anonymous>' make[1]: *** [libfreeswitch_la-switch_core_sqldb.lo] Fehler 1 make: *** [all] Fehler 2 Fehler = Error I did not find a solution anywhere in the internet. Any hints? Regards Sascha -- AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied --> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister Montabaur B14998 Bücher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de

2 2

FreeSWITCH OCF Resource Agent to run "sofia recover"
by Gavin Henry 14 Mar '11

14 Mar '11

Hi all, I saw this in Jan but gather the person is at the same point as us: http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/066729.… I'm putting together two test VirtualBox VMs which have pacemaker and DRBD running. The actual plan is FreeSWITCH, PostgreSQL, Apache and FusionPBX on the one box to start with in testing. FusionPBX (http://www.fusionpbx.com) will be running in full Multi-tenant mode as we've just sponsored the FusionPBX project to get the last bits done. Apache and Pg have resource agents already and FS will have its data as per the FS HA wiki docs pointing to a Pg database. The last piece is how to add in FreeSWITCH and run "sofia recover" "correctly". I'd rather not have a "hacked" together OCF resource agent for FS but a proper one that is in the FS git repo. I'll happily write this all up on the existing wiki pages, but for the multi-tenant parts you'll need to speak to the FusionPBX guys for help on that as you'll need commercial support from them for the setup. FYI we'll also be using LINBIT packages (http://www.linbit.com/en/products-services/linbit-cluster-stack-support/) for our new production platform for the stacked DRBD setup for off-site disaster recovery: http://www.drbd.org/users-guide/s-pacemaker-stacked-resources.html http://www.drbd.org/users-guide/s-three-way-repl.html Thanks, Gavin. -- http://www.suretecsystems.com/services/openldap/ http://www.surevoip.co.uk

1 0

exec app on originate
by Madovsky 14 Mar '11

14 Mar '11

when I do this : originate /sofia/gateway/blabla/9999999999 &app(hahaha) is the ap executed at early media or after answer ? Thanks

4 7

libss7 + asterisk woes!
by Anita Hall 13 Mar '11

13 Mar '11

Hi I am using Asterisk on Sangoma cards :( I know I should have used Digium but now the customer has these cards and I have to make them work with SS7 :( (And FreeSWITCH still does not support SS7 , is that right ?) Unable to receive incoming calls on chan_ss7, I have decided to give another shot to libss7 Could someone please share their configuration of dahdi/system.conf, chan_dahdi.conf, wanpipeX.conf with me for multiple signalling links ? I am especially confusing with the numbering schemes of CICs and signalling channels. I have signalling links on the first 2 spans, on 16th channel each. So , how do I number after the 2nd channel onwards. Is it 32-64 or 31-63 or what ? :) My current config is debian:/usr/src# cat /etc/dahdi/system.conf #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit #autogenrated on 2011-03-13 #Dahdi Channels Configurations #For detailed Dahdi options, view /etc/dahdi/system.conf.bak loadzone=us defaultzone=us #Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1> span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 mtp2=16 #Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2> span=2,2,0,ccs,hdb3,crc4 bchan=33-46,48-62 mtp2=47 #Sangoma A108 port 3 [slot:4 bus:5 span:3] <wanpipe3> span=3,0,0,ccs,hdb3,crc4 bchan=63-93 #Sangoma A108 port 4 [slot:4 bus:5 span:4] <wanpipe4> span=4,0,0,ccs,hdb3,crc4 bchan=94-124 #Sangoma A108 port 5 [slot:4 bus:5 span:5] <wanpipe5> span=5,0,0,ccs,hdb3,crc4 bchan=125-155 #Sangoma A108 port 6 [slot:4 bus:5 span:6] <wanpipe6> span=6,0,0,ccs,hdb3,crc4 bchan=156-186 #Sangoma A108 port 7 [slot:4 bus:5 span:7] <wanpipe7> span=7,0,0,ccs,hdb3,crc4 bchan=187-217 #Sangoma A108 port 8 [slot:4 bus:5 span:8] <wanpipe8> span=8,0,0,ccs,hdb3,crc4 bchan=218-248 #Sangoma A108 port 1 [slot:4 bus:7 span:9] <wanpipe9> span=9,0,0,ccs,hdb3,crc4 bchan=249-279 #Sangoma A108 port 2 [slot:4 bus:7 span:10] <wanpipe10> span=10,0,0,ccs,hdb3,crc4 bchan=280-310 #Sangoma A108 port 3 [slot:4 bus:7 span:11] <wanpipe11> span=11,0,0,ccs,hdb3,crc4 bchan=311-341 #Sangoma A108 port 4 [slot:4 bus:7 span:12] <wanpipe12> span=12,0,0,ccs,hdb3,crc4 bchan=342-372 #Sangoma A108 port 5 [slot:4 bus:7 span:13] <wanpipe13> span=13,0,0,ccs,hdb3,crc4 bchan=373-403 #Sangoma A108 port 6 [slot:4 bus:7 span:14] <wanpipe14> span=14,0,0,ccs,hdb3,crc4 bchan=404-434 #Sangoma A108 port 7 [slot:4 bus:7 span:15] <wanpipe15> span=15,0,0,ccs,hdb3,crc4 bchan=435-465 #Sangoma A108 port 8 [slot:4 bus:7 span:16] <wanpipe16> span=16,0,0,ccs,hdb3,crc4 bchan=466-496 debian:/usr/src# cat /etc/asterisk/chan_dahdi.conf ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2011-03-13 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A108 port 1 [slot:4 bus:5 span:1] <wanpipe1> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 ss7type=itu ;using the ITU variant ss7_called_nai=dynamic ;NAI for outgoing calls ss7_calling_nai=dynamic ;NAI for incoming calls ss7_internationalprefix=00 ;international prefix value for incoming calls ss7_nationalprefix=0 ;national prefix value for incoming calls ss7_subscriberprefix= ;subscriber prefix value for incoming calls ss7_unknownprefix= ;unknown prefix value for incoming calls ss7_explictacm=yes ;ACM is send as soon as call enters the dial plan...may not accepted yet though linkset=1 ;arbitrary name for this set of channels pointcode=13323 ;the point code for this system...aka SPC adjpointcode=12650 ;the point code for the system that we are signaling to... aka APC defaultdpc=12650 ;the point code for the system that the CICs will be negotiated with...aka DPC networkindicator=national_spare ;NI value for MTP3 cicbeginswith=1 ;the starting value of the CICs channel=2-31 ;the channels that are CICs sigchan=1 ;the signaling channel ;Sangoma A108 port 2 [slot:4 bus:5 span:2] <wanpipe2> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 ss7type=itu ;using the ITU variant ss7_called_nai=dynamic ;NAI for outgoing calls ss7_calling_nai=dynamic ;NAI for incoming calls ss7_internationalprefix=00 ;international prefix value for incoming calls ss7_nationalprefix=0 ;national prefix value for incoming calls ss7_subscriberprefix= ;subscriber prefix value for incoming calls ss7_unknownprefix= ;unknown prefix value for incoming calls ss7_explictacm=yes ;ACM is send as soon as call enters the dial plan...may not accepted yet though linkset=1 ;arbitrary name for this set of channels pointcode=13323 ;the point code for this system...aka SPC adjpointcode=12650 ;the point code for the system that we are signaling to... aka APC defaultdpc=12650 ;the point code for the system that the CICs will be negotiated with...aka DPC networkindicator=national_spare ;NI value for MTP3 cicbeginswith=1 ;the starting value of the CICs channel=2-31 ;the channels that are CICs sigchan=1 ;the signaling channel ;Sangoma A108 port 3 [slot:4 bus:5 span:3] <wanpipe3> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>63-77,79-93 ;Sangoma A108 port 4 [slot:4 bus:5 span:4] <wanpipe4> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>94-108,110-124 ;Sangoma A108 port 5 [slot:4 bus:5 span:5] <wanpipe5> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>125-139,141-155 ;Sangoma A108 port 6 [slot:4 bus:5 span:6] <wanpipe6> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>156-170,172-186 ;Sangoma A108 port 7 [slot:4 bus:5 span:7] <wanpipe7> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>187-201,203-217 ;Sangoma A108 port 8 [slot:4 bus:5 span:8] <wanpipe8> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>218-232,234-248 ;Sangoma A108 port 1 [slot:4 bus:7 span:9] <wanpipe9> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>249-263,265-279 ;Sangoma A108 port 2 [slot:4 bus:7 span:10] <wanpipe10> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>280-294,296-310 ;Sangoma A108 port 3 [slot:4 bus:7 span:11] <wanpipe11> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>311-325,327-341 ;Sangoma A108 port 4 [slot:4 bus:7 span:12] <wanpipe12> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>342-356,358-372 ;Sangoma A108 port 5 [slot:4 bus:7 span:13] <wanpipe13> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>373-387,389-403 ;Sangoma A108 port 6 [slot:4 bus:7 span:14] <wanpipe14> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>404-418,420-434 ;Sangoma A108 port 7 [slot:4 bus:7 span:15] <wanpipe15> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>435-449,451-465 ;Sangoma A108 port 8 [slot:4 bus:7 span:16] <wanpipe16> switchtype=national context=from-pstn group=0 echocancel=no signalling=ss7 channel =>466-480,482-496 debian:/usr/src# cat /etc/wanpipe/wanpipe1.conf #================================================ # WANPIPE1 Configuration File #================================================ # # Date: Wed Dec 6 20:29:03 UTC 2006 # # Note: This file was generated automatically # by /usr/local/sbin/setup-sangoma program. # # If you want to edit this file, it is # recommended that you use wancfg program # to do so. #================================================ # Sangoma Technologies Inc. #================================================ [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 5 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 430 LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 0 TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS Blue Alarm and keep line down #wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode #wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode TDMV_HW_DTMF = NO # YES: receive dtmf events from hardware TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events from hardware HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation enabled with nlp (default) # OCT_SPEECH: improves software tone detection by disabling NLP (echo possible) # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions. HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of incoming media (must have hwdtmf enabled) HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the line - could break fax HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo cancelation HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software tone detection (possible echo) HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal [w1g1] ACTIVE_CH = ALL TDMV_HWEC = NO MTU = 8

2 1

invalid format after speex encoding
by Madovsky 13 Mar '11

13 Mar '11

I tried to convert vm 16000hz wav sound files to spx but FS can't read them: "invalid format" is there any specail encoding settings to make them work ? Thanks

1 0

unknown event 8: 101 NAT detected
by Johan Helsingius 13 Mar '11

13 Mar '11

I have a lot of problems getting my SIP provider connections to work. I am using 3 separate SIP providers (for different destinations/services), and sometimes things work and sometimes they don't. Registrations work OK, but often there is no audio, or the call establishment fails. Makes me suspect a network/NAT issue. Something that I do worry about is the log messages saying "sofia.c:995 nua_i_outbound: unknown event 8: 101 NAT detected". Julf

1 1

question about faxes
by Boris Kovalenko 13 Mar '11

13 Mar '11

Hello! Here is a small topology description: FaxA -- ClientA -- (internet) -- myFS -- ClientB -- FaxB ClientA - somewhere in the world, with unknow equipment myFS - FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300) managed by me. As I need to process DTMF from clients, proxy_media=false ClientB - any of my clients with some type of equipment Voip<->FXS conversion FaxA/B - Faxes May someone tell me please what configuration should I do to permit my clients send and receive faxes? The main problem is that I really don't know when and on what number the user want to use fax machine. I have read mod_spandsp wiki but it isn't so clear. Is this enough to add this applications before any bridge?: <action application="set" data="fax_enable_t38=true"/>  <action application="set" data="fax_enable_t38_request=true"/>  <action application="set" data="execute_on_answer=t38_gateway self"/>  -- С уважением, Борис Коваленко ЗАО "Тагнет" (3435) 494991

1 0

Another beginner question: garbled voicemail
by Johan Helsingius 13 Mar '11

13 Mar '11

Hi! For some reason the recorded voicemail messages come out as random noise. All the system messages sound fine, it's just that the recorded message is pretty much random garbage. Any hints for where to start looking? Julf

3 9

mod_callcenter not bridging call to agent (callback)
by guru singh 13 Mar '11

13 Mar '11

Hi, I am trying to get mod_callcenter working. When the callcenter extension is dialed, the caller keeps listening to moh until it times out. The call is not being sent to agents. call debug log and other info here http://pastebin.freeswitch.org/15665 What am i missing? Let me know if anything else is required. thanks guru

1 0