February 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

execute_on_preanswer or execute_on_pre_answer?
by Fabio Mierlo 14 Feb '14

14 Feb '14

Hello, I noticed that execute_on_pre_answer variable is being mentioned as execute_on_preanswer in the whole wiki. Regards, Fabio Mierlo

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Re: [Freeswitch-users] Freeswitch not putting external IP in SDP and From Header when dialing out SIPURI by external profile
by Andrew Paul 13 Feb '14

13 Feb '14

Hai brian, I given my router ip in ext-sip-ip and ext-rtp-ip. Ext-rtp-ip = router-ip Issue is fs not putting profile->extrtpip because its fail in nat checking. Regards Andrew On Sunday, February 9, 2014, Brian West <brian(a)freeswitch.org> wrote: > What values do you have in ext-sip-ip and ext-rtp-ip? > -- > Brian West > brian(a)freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > On Feb 9, 2014, at 9:09 AM, Andrew Paul <andrew.paul85(a)gmail.com> wrote: > >> Hi all, >> >> >> ------------------FREESWITCH--------------(bridge=sofia/external/sip:outbound-opensip.com:5060 )-----------OPENSIP---------------------- >> >> I am facing NAT problem when i am dialing SIP URI via sip profile. When i am dialing the SIP uri with Gateway trunk NAT is working properly. >> >> Dialing SIP URI >> >> <action application="bridge" data="sofia/external/sip: outbound-opensip.com:5060" >> >> In external profile setting I made the setting for ext-rtp-ip and ext-sip-ip. In freeswitch cli "sofia status profile external" i am able to see the external rtp and sip ip correctly. >> >> But when freeswitch sending INVITE external IPs are not set in SDP and FROM Header, instead it is using local IP. When i tried to debug the problem in switch_core_media.c freeswitch failed to check the NAT settings. >> >> switch_core_media.c >> >> SWITCH_DECLARE(switch_status_t) switch_core_media_choose_port(switch_core_session_t *session, switch_media_type_t type, int force) >> >> /* Check if NAT is detected */ >> if (!zstr(smh->mparams->remote_ip) && switch_core_media_check_nat(smh, smh->mparams->remote_ip)) { >> /* Yes, map the port through switch_nat */ >> /* Code */ >> } else { >> /* No NAT traversal required, use the profile's rtp ip */ >> use_ip = smh->mparams->rtpip; >> } >> >> Here remote IP is NULL always when i am dialing URI with external profile. When I tried to make call by "sofia/gateway/trunk/sip: outbound-opensip.com:5060" i can see the remote ip as "oubound-opensip.com" and its able to external IP in SDP. >> >> Can any body tell how to solve this problem ? Is there any configuration to set relam in dialplan ? >> >> I am using freeswitch version 1.5.6. >> >> Hopping the earliest response. >> >> Regards >> >> Andrew > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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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blocked 10 secs in hangup
by Javier Menendez 13 Feb '14

13 Feb '14

I have very weird behaviour when some calls hangup, freeswitch freezes for exactly 10 seconds as the log shown below. I don't see any trace of error but for 10 seconds even the audio is not delivered, I guess is some timeout related issue but I don't know where to look any clue? 2014-02-12 18:39:25.668742 [DEBUG] switch_core_state_machine.c:579 (sofia/external/spyU0lQL2Zyb210ZG0tMDAyMDNiYjQ=(a)10.10.20.6) State DESTROY going to sleep 2014-02-12 18:39:35.668734 [DEBUG] switch_core_state_machine.c:48 rtmp/default/chanspy_001-1392227350.2513709 Standard HANGUP, cause: NORMAL_CLEARING

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ded channels
by Jason Holden 13 Feb '14

13 Feb '14

Hi, When reviewing my demo fs the other day saw that fs was hanging channels in ringing state etc. Was this a bug with v1.2.20 and corrected in 1.2.21? Just update to v.1.2.21 this morning.

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esl, send_dtmf, too early ESL_EVENT_CHANNEL_EXECUTE_COMPLETE
by Marcin S 13 Feb '14

13 Feb '14

Hello, In my simple C application I want to send dtmfs. This is quite easy - you just call esl_execute("send_dtmf", "1234", uuid); and it works. However, ESL returns ESL_EVENT_CHANNEL_EXECUTE_COMPLETE way to fast - even before it starts dtmfs generation! How can I know, when dtmfs were really generated? So far, the only method is to issue some sleep(), but it looks like big workaround.

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routing incoming SMS?
by Peter Villeneuve 13 Feb '14

13 Feb '14

Hi, I'm experimenting with GSMopen and I'm having trouble doing something which is likely pretty simple. I have enabled mod_sms and GSMopen in FS. I can send SIP simple messages between 2 registered peers OK. I can see that incoming GSM SMS messages arrive in FS as expected, but I can't seem to get them converted into SIP SIMPLE format and routed to the peer (1000). Once they arrive in FS, I can see in the log: 2014-01-30 20:00:54.840999 [INFO] mod_sms.c:336 Processing text message +4412398746->gsm01 in context default Chatplan: gsm01 parsing [default->basic p2p] continue=true Chatplan: gsm01 Regex (PASS) [basic p2p] to(gsm01) =~ /^(.*)$/ break=on-false Chatplan: gsm01 Action send() My chatplan has: <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="basic p2p" continue="true"> <condition field="to" expression="^(.*)$">  <action application="send"/> </condition> </extension> </context> </include> And in my gsmopen.conf.xml: <configuration name="gsmopen.conf" description="GSMopen Configuration"> <global_settings> <param name="debug" value="8"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="hold-music" value="$${moh_uri}"/> <param name="destination" value="1000"/> </global_settings> I believe I'm doing something wrong in the chatplan. Is there any transfer action like in the regular XML dialplan? I tried creating in the default dialplan the following, hoping that it would be similar to sofia calls and transfer the incoming SMS to peer 1000, but no dice. <include> <extension name="sms_inbound"> <condition field="destination_number" expression="^(gsm01)$"> <action application="transfer" data="1000 XML default"/> </condition> </extension> </include> So, basically, how can I route the incoming SMS to peer 1000? Thanks

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ignore_display_updates blocks display name for IVR
by mehmet ozisik 13 Feb '14

13 Feb '14

Hi, I am using FreeSWITCH-mod_sofia/1.5.8b+git~20140204T065630Z~5228e02b19~64bit. To block to get display update after connected to IVR , I have added <X-PRE-PROCESS cmd="set" data="ignore_display_updates=false"/> to vars.xml and FS does not send update anymore. But this time on the phone screen, "Outgoing Call" is written instead of IVR. I have checked the code find the place what causes this, but I wonder this is related with this variable somehow. I would like to ask you that is there any way to send IVR string as displayname? Remote-Party-ID: "Outgoing Call" <sip:IVR@192.168.169.169 >;party=calling;privacy=off;screen=no regards Mehmet

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libsofia_sip_ua_static won't compile
by Andre 13 Feb '14

13 Feb '14

When trying to compile the latest stable branch in VS 2012, release, 64 bit libsofia_sip_ua_static has over 807 Errors. So many it's hard to know what is wrong. Anyone have a clue what's wrong? Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory ..\libs\sofia-sip\libsofia-sip-ua\sip\sip_extra.c 45 1 libsofia_sip_ua_static Error 1845 error C2059: syntax error : ')' ..\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h 188 1 libsofia_sip_ua_static Error 758 error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory ..\libs\sofia-sip\libsofia-sip-ua\sip\sip_extra.c 45 1 libsofia_sip_ua_static

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xml cdr problems
by Jason Holden 12 Feb '14

12 Feb '14

Hi, I have a fs that is running an older master build of 1.3.0 and I'm noticing speratic problems with cdr postings where it doesn't post, or fields are missing. Is anyone awhere of issues with xmlcdr around that release? Is their a way I can debug this in the fs cli? Currently I've looked through my logs and am not seing any xmlcdr errors. Thanks.

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FYI - sql quoting issue with mod_skinny configs
by Nathan Neulinger 12 Feb '14

12 Feb '14

FS-6220 fixed with c65f2884747bfd4c0a97d540620a58596eb8f595 If you were taking any special steps to double single quotes inside of mod_skinny configs (such as for speed dials) that is now corrected in the code. None of the mod_skinny queries were protecting sql quotes. All of those cases have been corrected in both v1.2stable and master. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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