- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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NAT Help needed
by Simon Leck 26 Dec '08

26 Dec '08

Hi Everybody I am a newbie in this. I have managed to setup freeSwitch but I am unable to resolve the NAT issue. Hope somebody out there can furnish me with guidance. So far I have manage to use "Use Stun server to resolve registration problem regarding client behind NAT device but behind two layers, NAT devices still have problem in registration. Thanks Everybody for your kind assistance. Simon Email: simon0922(a)gmail.com

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Limiting port for OpenMRCP
by mark morreny 26 Dec '08

26 Dec '08

Hi, I have a dev version of a MRCP-supported TTS, but it can only allow 1 port to connect. Does OpenMRCP has a way to limit the number of port to connect and queue the rest of the request somehow? Thanks for your suggestion and help. Mark

1 0

Setting up port audio for incoming/outgoing calls
by Jason White 26 Dec '08

26 Dec '08

On the wiki, an example of a port audio configuration is given that involves creating a Sip gateway on localhost. As I couldn't get this to work (apparently due to the external profile's detection of NAT), I thought I would try an alternative approach. I am modifying the default dial plan here. At some point I'll probably just rewrite it anyway. I have created a user in the directory for extension 1020. For outbound calls, in default.xml, I have the following: <extension name="portaudio" continue="true"> <condition field="source" expression="mod_portaudio"> <action application="set_user" data="1020@$${domain}"/> <action application="set" data="effective_caller_id_number=1020"/> </condition> </extension> The log shows that the set_user is executed, as is the set effective_caller_id_number (the latter shouldn't be necessary, unless I'm misunderstanding). However, running show channels after making a call from the portaudio device still shows the user name and caller id as FreeSWITCH,0000000000 Also, when I try to call a local extension from the audio device, I get the following in the logs, and the call is terminated. I've checked the code, and clearly the failure to open the file is the cause of the termination. The Sip phone on the extension rings once and then it receives the cancellation from FreeSWITCH. 2008-12-25 10:29:59 [DEBUG] switch_ivr_originate.c:1313 switch_ivr_originate() P lay Ringback File [local_stream://moh] 2008-12-25 10:29:59 [ERR] mod_local_stream.c:308 local_stream_file_open() Unknow n source moh 2008-12-25 10:29:59 [ERR] switch_ivr_originate.c:1322 switch_ivr_originate() Err or Playing File 2008-12-25 10:29:59 [DEBUG] switch_core_codec.c:122 switch_core_session_set_read _codec() Restore original codec. 2008-12-25 10:29:59 [NOTICE] switch_ivr_originate.c:1560 switch_ivr_originate() Hangup sofia/internal/sip:1000@192.168.0.4:2048;line=mxyv04us [CS_CONSUME_MEDIA] [NO_ANSWER] 2008-12-25 10:29:59 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup( ) Send signal sofia/internal/sip:1000@192.168.0.4:2048;line=mxyv04us [KILL] Any hints would be welcome. There is no urgency, of course, as I'm doing this for fun and out of interest. Happy holidays to all on the FreeSWITCH list.

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Behind a router
by kriko 25 Dec '08

25 Dec '08

Hello! I installed freeswitch on different configuration, but can't get registration working. So my phone (192.168.10.2) is now behind a wireless router (192.168.10.1), ethernet port (192.168.0.5) is attached to eth1 (192.168.0.1). Fs is running on that machine (it has eth0 - net and eth1 - inside lan), sofia shows: http://pastebin.com/m6f349c32 and registration fails: http://pastebin.com/m2916e20d How could I fix that? So when I would call 1000(a)212.235.180.41 would ring my phone at 192.168.10.2? I've setup port forwarding on port 5060 between 192.168.10.2 and 192.168.0.1 both ways. --

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