December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

onInputCallback unstable in Python
by Adam Wilt 29 Dec '08

29 Dec '08

I'm using builds 10724 and 10914 to place an outbound call from the FreeSwitch console and use the onInputCallback functionality. My goal is to get mod_vmd working for me. When I run my script and press a touchtone to invoke the callback function, I get some unstable behavior; sometimes it works fine, sometimes I get a core dump (send me an email at wiltingtree(a)gmail.com if anybody wants the core dump), sometimes it gives me the following error: TypeError: onInputCallback() takes exactly 3 arguments (0 given) In the documentation onInputCallback() takes 3 arguments, and I don't see how it would be very useful with zero. Here is a test script I put together which shows this behavior: import os from freeswitch import * def onInputCallback(session, what, obj): consoleLog("INFO","IM IN THE CALLBACK!\n") return("continue") def fsapi(session, stream, env, args): consoleLog("INFO","Hello there!!!\n") session = Session("{ignore_early_media=true}sofia/gateway/gafachi/1xxxxxxxxxx") session.sleep(500) session.setInputCallback(onInputCallback) session.streamFile("/root/intro.wav") consoleLog("info","Bye!\n") session.hangup() return(session)

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Multiple context without using directory
by Ron McCarthy 29 Dec '08

29 Dec '08

Hi all, I would like to setup FS to have many context, basically we just want to switch calls, and since we have been using Asterisk we want to keep the context names the same, and well it's easier it seems to me. This is for termination only, so just sending calls out, I would think FS can do it the same way Asterisk does, where if the IP matches then it will use those settings with that gateway and diaplan since the context ties them all together. No matter what I try the context always goes to default, I even took off all the ACL's for internal and external profiles and tried gateways in each, still no luck, i can't get around the default context. We are going to be using IP based auth only, so no user/pass's ever. Any help on this would be great, I have searched and searched and everyone just uses the default context, but I am trying to avoid that as the diaplan will get quite long as I will have to all kinds of crazy matching, when a context is much simpler, and I would think faster then using lots of dialplan logic. Thanks for any help on this! Ron

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Freeswitch/Sofia configuration problem
by Laurent Fabre 28 Dec '08

28 Dec '08

Hello, I've been trying to figure out for a few days why my freeswitch instance suddenly become insensitive to SIP packets without any warning. What usually happen is the following : 1) start just fine in foreground mode and no errors 2) wait anywhere between 2 seconds and 20 minutes 3) Sofia suddenly decide to reload everything for some reason 4) Sofia start processing SIP packets 5) work for an hour or so 6) Sofia suddenly decide to reload everything for some reason 7) become unresponsive again 8) goto 2 Both interfaces have public IP addresses assigned in a static manner (no DHCP). I can see the SIP UDP & TCP requests comming from the phones on several sites on the wire. The SIP TCP requests get RST in reply which is mean :( There was a point in my setup where it would not happen but since I'm new to freeswitch I'm having an hard time backtracking. I was wondering if iproute/tc and iptables were the culprits but I flushed everything (even rebooted without loading the rules) and it still doesn't work. I thought some database was corrupt so I shutdown'd freeswitch and delete his db folder, no effect. My server runs Debian 4.0etch for amd64, built freeswitch from SVN trunk. Any pointers, help, cure against headaches would be great :) Regards, Laurent

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lua call to stop_record_session - INVALID COMMAND
by Frank ＠ Impact 28 Dec '08

28 Dec '08

I was trying to stop a session record from lua but when I try I get a "Result is INVALID COMMAND!" I am calling this lua script with <action application="set" data="api_hangup_hook=lua proc.lua .."/> so by the time the lua is called, someone has hungup one of the legs. In the lua script I am using this to try to end the record session to the wav file so it gets closed and so I can convert it to mp3 right away and a few other things in the lua script... apicmd = "stop_record_session"; apiarg = recordfile; res = api:execute(apicmd,apiarg); but that is when I get the INVALID COMMAND on the freeswitch console. Is there a proper way to do this from lua?

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Hard limit on RTP sessions
by Juan Backson 27 Dec '08

27 Dec '08

Hi, Is there any hard limit set on the number of RTP sessions for Freeswitch? I am seeing freeswitch start sending out BYE after the number of RTP session reaches 3000. This problem happens even when the machine utilization is still low. Does anyone know what is wrong? Thanks, JB

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Re: [Freeswitch-users] api_hangup_hook and bash
by Michael S Collins 26 Dec '08

26 Dec '08

I'm pretty sure that this is doable. Could you give us a hint as to what arguments you want to send? For example, do you have one or more channel variables you'd like to pass to the shell script? -MC Sent from my iPhone On Dec 23, 2008, at 6:25 PM, Jason White <jason(a)jasonjgw.net> wrote: > Frank @ Impact <frank(a)impactfax.com> wrote: >> Can this command be used to run a bash script? > > Based on information at the wiki, this should be possible; use the > system > command. > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system > > > _______________________________________________ > 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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Re: [Freeswitch-users] group_confirm seems to be broken
by Adnan Barakat 26 Dec '08

26 Dec '08

Hi all, I've just updated from r10000 to the latest trunk (as I needed mod_http), and group_confirm seems to have broken after the update. Now the first 2-5 seconds of the file is played very quickly at poor quality, then the end of the file plays fine. Here is the relevant part of the dialplan; <action application="set" data="timer_name=soft"/> <action application="set" data="group_confirm_key=exec"/> <action application="set" data="group_confirm_file=playback /somefile.al"/> The last time this happened timer_name=soft fixed the problem, but doesn't seem to make any difference in this case. Thanks Adnan

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another switch_ivr_set_user() can't find user
by John Wehle 26 Dec '08

26 Dec '08

>> a) Should sip_auth_realm be set by FreeSWITCH to the value associated >> with force-register-domain > > You have to remember the default assumes a lot. You go to changing > things you have to then change the way things are assumed. I appreciate that. Let me ask the question slightly differently. sofia_reg_parse_auth contains the following logic: if (!switch_strlen_zero(profile->reg_domain)) { domain_name = profile->reg_domain; } else { domain_name = realm; } where profile->reg_domain is set from force-register-domain. It then calls switch_xml_locate_user using domain_name. It looks like force-register-domain is intended to make FreeSWITCH believe that the user is in domain specified by force-register-domain. Later there's: switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM, "sip_auth_realm", realm); switch_event_add_header_string(*v_event, SWITCH_STACK_BOTTOM, "domain_name", realm); Shouldn't the add_header for domain_name contain the value for the actual domain used to locate the user? And ideally shouldn't the rest of FreeSWITCH (including examples intended to get you started) work in the same fashion for consistency sake (i.e. when trying to locate a user reference the domain used by sofia_reg_parse_auth to locate the user instead of blindly using sip_auth_realm)? My thought is if sofia_reg_parse_auth set things up properly, then the rest of FreeSWITCH shouldn't know or even care that force-register-domain is in use ... it should be as if the VoIP phone had in fact registered using the domain specified by force-register-domain. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Need help with openmrcp setup
by mark morreny 26 Dec '08

26 Dec '08

Hi I tried to setup mod_openmrcp according to wiki, but I am getting the following error: 2008-12-27 00:40:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]! 2008-12-27 00:40:23 [ERR] switch_ivr_play_say.c:1848 switch_ivr_speak_text() Invalid TTS module! Here is my dialplan <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="dialplan" description="voicemail dialplan"> <context name="public"> <extension name="test9"> <condition field="destination_number" expression="^(.*)$"> <action application="speak" data="openmrcp:openmrcp-v2|Susan|Hello World"/> </condition> </extension> </context> </section> </document> I checked that mod_openmrcp.so is compiled ok: [root@localhost bin]# ls ../mod/mod_openmrcp.so -al -rwxr-xr-x 1 root root 4981560 Dec 26 19:56 ../mod/mod_openmrcp.so [root@localhost bin]# Here is what I have in the openmrcp xml config: [root@localhost freeswitch]# cat conf/mrcp_profiles/openmrcp-v2.xml <include> <profile name="openmrcp-v2" description="OpenMRCPv2 Server"> <param name="proto_version" value="2"/> <param name="client_ip" value="127.0.0.1"/> <param name="client_port" value="8062"/> <param name="server_ip" value="192.168.1.109"/> <param name="server_port" value="5060"/> <param name="rtp_port_min" value="11000"/> <param name="rtp_port_max" value="12000"/> </profile> </include> Could someone help me out? I would greatly appreciate any help. Thanks, Mark

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strange error while running stress testing
by Juan Backson 26 Dec '08

26 Dec '08

Hi I am getting the following strange error while running stress test on freeswith. When the number of sessions reaches 3000, I get the following error: 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:40 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate() Error[[error near line 1]: root tag missing] 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate() Error[[error near line 1]: root tag missing] 2008-12-24 15:37:41 [ERR] switch_xml.c:1476 switch_xml_locate() Error[[error near line 1]: root tag missing] 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate() Error[[error near line 1]: root tag missing] 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate() Error[[error near line 1]: root tag missing] 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate() Error[[error near line 1]: root tag missing] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Timeout] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Timeout] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Timeout] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Timeout] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Timeout] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Timeout] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Timeout] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Timeout] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate() Error[[error near line 1]: root tag missing] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate() Error[[error near line 1]: root tag missing] 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate() Error[[error near line 1]: root tag missing] 2008-12-24 15:37:42 [ERR] sofia_glue.c:2056 sofia_glue_activate_rtp() AUDIO RTP REPORTS ERROR: [Bind Error!] 2008-12-24 15:37:42 [ERR] sofia.c:3020 sofia_handle_sip_i_state() RTP Error! 2008-12-24 15:37:42 [ERR] switch_xml.c:1476 switch_xml_locate() Error[[error near line 1]: root tag missing] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] 2008-12-24 15:37:42 [ERR] sofia_glue.c:490 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478 [Remote Address Error!] Could someone help me out? What do those errors mean? Thanks in advance for all your help. JB

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