March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

Is external_rtp_ip=auto still valid?
by Mario G 12 Mar '11

12 Mar '11

The wiki states auto is a valid option under Sofia configuration but it is not mentioned in the vars.xml. Is this still valid? I need and way to make this work without uPNP. Thanks, Mario G

1 0

inbound dtmf issue after upgrade
by Kevin Raison 12 Mar '11

12 Mar '11

I upgraded a freeswitch setup from 1.0.4 to 1.0.6 yesterday, and also upgraded my wanpipe drivers from 3.4.9 to 3.5.10. I am using openzap with a sangoma card. Before the upgrade, the following stanza would correctly send a caller to a mod_event_socket-based application that I have running on another machine and that caller's dtmf would flow through to the app: <extension name="Kevin_test"> <condition field="destination_number" expression="^XXXXXXXXXX$"> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="bridge" data="sofia/internal/5000(a)prod.XXXXXXX.com"/> </condition> </extension> After the upgrade, everything but inbound dtmf works as it should, but the only way that I can get inbound dtmf to work is by adding <action application="start_dtmf"/> to the stanza. This is not ideal and I would like to go back to not using in-band dtmf if possible. In my log files, I see absolutely no dtmf being registered when a caller calls in. However, if someone calls out from a desk phone and then transfers the person that they called to sofia/internal/5000(a)prod.XXXXXXX.com, dtmf works just fine. Any ideas? Thanks! Kevin

2 3

How to timeout att_xfer?
by Boris Kovalenko 12 Mar '11

12 Mar '11

Hello! Is there a possibility to timeout att_xfer application? For example -> A calls B, B does att_xfer to C, but C doesn't answer for a 10 seconds so B is bridged again to A (of course if B is still off hook). -- С уважением, Борис Коваленко ЗАО "Тагнет" (3435) 494991

2 6