August 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Can a chat message be sent to a cell phone with FS?
by Merle J. Ebbert 27 Aug '09

27 Aug '09

Can a chat message be sent to a cell phone with FS? Thanks, Merle

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Call exits after 120 seconds with hangup cause
by Dennis 27 Aug '09

27 Aug '09

hi, we are using fs for different services, but we never used it, to connect sip-phones directly with fs. now we want to do so, but we encounter big problems. everything works fine, but after 120 seconds fs hangs up with the hangup cause RECOVERY_ON_TIMER_EXPIRE. it seems that this has something to do with nat problems (i read about it in nabble), but we can not figure out, how to fix the problem. we are not using stun, because all ip-addresses are static. we opened our firewall for our ip-addresses and opened the following ports: 5060 - 5091 TCP / UDP 10000 - 32767 UDP we are using a quite actual version of fs - the revision ist 13783. we set up a profile in the internal.xml and played a lot with different settings. we can talk to each other, make outbound calls and receive calls - but we can not figure out, how to avoid the hangup after 120 seconds. thanks a lot for your help dennis

5 19

mod_voicemail email template variables
by Nick Lemberger 27 Aug '09

27 Aug '09

Is there a way to use dialplan variables in the email that gets sent with the voicemail attachement. I tried using some but nothing seems to show up, I'm guessing it's a different channel or something... Any ideas? Thanks, -Nick

2 3

Calls from registered gateway try to lookup Directory
by Peter P GMX 27 Aug '09

27 Aug '09

I have found a strange thing in my FS installation, FS is registered via a Gateway to an external provider (QSC) in the external context. But when a call is coming in, FS does not seem to go to any context, but tries to lookup the user, as I receive the following message 2009-08-27 18:44:23.287782 [WARNING] sofia_reg.c:1773 Can't find user [026xxxxxxxxx@my.domain@my.domain] You must define a domain called 'my.domain' in your directory and add a user with the id="026xxxxxxxxx(a)my.domain" attribute and you must configure your device to use the proper domain in it's authentication credentials. I learnt that a call from an external gateway should go to the public context. But (in CLI debug mode) there are no other messages, except the 3 lines above. What am I doing wrong? Best regards Peter Here the invite message. INVITE sip:gw+gw_xxxxxxxxxx@xx.xxx.xx.xxx:5080;transport=udp SIP/2.0. Via:SIP/2.0/UDP 62.206.3.xxx;branch=z9hG4bK-BroadWorks.as1-xx.xxx.xx.xxxV5080-0-778271239-1616003581-1251392025611-. From:"0xxxxXXXXXX"<sip:0xxxxXXXXXX@62.206.3.xxx;user=phone>;tag=1616003581-1251392025611-. To:"Me"<sip:026xxxxxxxxx@my.domain>. Call-ID:BW185345611270809356816303@62.206.3.xxx. CSeq:778271239 INVITE. Contact:<sip:62.206.3.xxx:5060>. Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE. Accept:multipart/mixed,application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp. Supported:. Max-Forwards:20. Proxy-Authorization:DIGEST cnonce="fyvqi2pf",qop=auth,uri="sip:gw+gw_026xxxxxxxxx@xx.xxx.xx.xxx:5080;transport=udp",realm="my.domain",username="026xxxxxxxxx(a)my.domain",nonce="21bbe70c-932a-11de-b94d-bbade892ded3",algorithm=MD5,response="6d39a2546a4aa9a1fc39e2dc07c1e934",nc=00000001. Content-Type:application/sdp. Content-Length:344. . v=0. o=BroadWorks 1271473 1 IN IP4 87.234.9.178. s=-. c=IN IP4 87.234.9.178. t=0 0. m=audio 18534 RTP/AVP 8 0 2 99 18 110. a=rtpmap:99 G726-24/8000. a=rtpmap:110 X-NSE/8000. a=fmtp:110 192-194,200-202. a=X-sqn:0. a=X-cap: 1 audio RTP/AVP 110. a=X-cpar: a=rtpmap:110 X-NSE/8000. a=X-cpar: a=fmtp:110 192-194,200-202. a=X-cap: 2 image udptl t38.

1 1

sofia profile external register gwname via XML-Curl?
by Peter P GMX 27 Aug '09

27 Aug '09

Hello, I am using XML-Curl to handle the configuration of freeswitch When I try to register a gateway via event-socket with sofia profile external register <gw-name> I receive back "invalid gateway". After reload mod_sofia the gateway is there. Question: Does this command work with xml-curl or only with local files?? At least I see no xml-curl request when grepping network traffic. Best regards Peter

1 1

FXO and analogue phones
by Merul Patel 27 Aug '09

27 Aug '09

I have a Freeswitch setup working on an Alix embedded platform in conjunction with a USB FXO device from Sangoma. My goal is to be able to either answer incoming calls on a softphone or on a POTS handset elsewhere in the building, and to also be able to make outgoing calls from either. For clarity, the analogue line has two physical extensions, one connected to the POTS and the other to the FXO. I can make and receive calls fine, but have problems when the call is answered on the POTS handset. Here is the dialplan I initially used in /opt/freeswitch/conf/ dialplans/public/01_incoming.xml: <include> <extension name="public_did"> <condition field="${strftime(%w)}" expression="^(\d)$">  <action application="sleep" data="23000"/> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="1001 XML default"/> </condition> </extension> </include> It's pretty basic, and if the softphone is not registered or does not answer then the call goes to voicemail. However the call will always go to voicemail, and the voicemail application will begin to execute after the call has been answered on the POTS handset. I've been trying to make the dialplan more useful, by having it ring the softphone immediately, and only transfer the call to the voicemail application if the line is still ringing. I'm in the UK, hence my choice of frequencies in the tone_detect application: <include> <extension name="public_did"> <condition field="${strftime(%w)}" expression="^(\d)$">  <action application="set" data="call_timeout=23"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/internal/1001%$$ {domain}"/> <action application="sleep" data="23000"/> <action application="tone_detect" data="ring 400,450 r +5000 set RING=true"/> <action application="transfer" data="public_answer_and_email"/> </condition> </extension> <extension name="public_answer_and_email"> <condition field="RING" expression="true"> <action application="answer"/> <action application="voicemail" data="default $${domain} 1001"/> </condition> </extension> </include> Unfortunately, this is not working, and the logs are not yielding anything the is helpful to me. Is my use of the tone_detect application and the basic dialplan correct? Merul

2 1

fifo question
by Juan Backson 27 Aug '09

27 Aug '09

Hi, Does anyone know the purpose of fifo_orbit_announce? When does fifo_orbit_announce get played? Thanks, JB

2 1

Question about presence
by Dennis 27 Aug '09

27 Aug '09

hi, we have set manage-presence = true to see, who is talking on the phone and who is free. everyone here has his own snom-voip phone with 12 led-lights. incoming calls are handled in groups. this means, if someone is ringing, all voip-phones of one group are ringing and one can see all led-lights flashing. if someone answers the phonecall, the led of this person (p1) is on on everybodys phone, so everybody can see, that the person is talking. now, if another phonecall comes in, all led-lights are flashing - including the led of the person who is talking. if another person (p2) answers the second phonecall, the led of p2 is on, but the led of p1 is off, although p1 is still talking. is there something we can do about this? thanks & kind regards dennis

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fscore mutex locking question
by Benedikt Fraunhofer 27 Aug '09

27 Aug '09

Hello *, while looking at the code i came across a region of code which is unclear to me regarding locking issues. One example is switch_ivr_broadcast in switch_ivr_async.c. This should be the function called by uuid_broadcast() and others. in line 2341 it tries to queue an event to the bleg if it has to... ----- . if ((flags & SMF_ECHO_BLEG) && (other_uuid = switch_channel_get_variable(channel, SWITCH_SIGNAL_BOND_VARIABLE)) . . && (other_session = switch_core_session_locate(other_uuid))) { --- switch_core_session_locate() does a "readonly trylock" on the channel mutex returning NULL if it's unable to aquire the lock, which brings up the following question: If some other thread is currently holding a writelock on the channel, the broadcast is not queued and not retried at a later time at all? I guess it's pretty easy to cause some unexpected behaviour using some endless loop calling "uuid_setvar" or some other race condition where the channel-mutex is write-locked while calling uuid_broadcast (eg. uuid_media?). Could this lead to a problem in "real live" scenarios, or are there other countermeasures despite "chances are one in a million that you hit that small time frame"? thx in advance Beni.

2 1

Get the Session State
by delianSPAM 27 Aug '09

27 Aug '09

Hello Everybody! How do you get the call state in Python? I have tried: . session.answer() state_result=str(session.getVariable("state")) console_log("debug",state_result) . But it returns: "None" Thank you! Best Regards, Delian Tashev

2 1