May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

sip response "101 dialing" - problems
by Anestis Mavro 06 May '12

06 May '12

After updating to git (adfe462) this Friday I noticed problems with outgoing calls from Linksys PAP2T and old 1001 devices. It looks like these devices can't handle the new SIP response "101 dialing" which is new (I don't know when it was introduced). As soon as you dial you get a "busy tone" but the call continues and after a while it rings the other end. Does anybody know how to disable this SIP response? (It comes after "100 trying") Thank you

2 1

FreeTDM and FXO Call Disconnect Supervision in Australia
by efs.tdw＠recursor.net 06 May '12

06 May '12

I have searched the archives (and indeed the Internet at large) and everything I found pointed me to the perfect solution to all my problems: get ROIC (Reverse On Idle Condition) enabled on my phone line. Sadly further investigation revealed that Telstra (who provides most PSTN services in Australia, including mine) has decided to discontinue offering ROIC as of 2008: > Line reversals were withdrawn from new sales June/July this year [2008] and will be removed from existing services June/July next > year [2009]. (http://forums.whirlpool.net.au/archive/1076769) Let me describe the kinds of problem I am observing: 1. Remote party calls, waits for 5 seconds, then hangs up before I have answered. My phones go on ringing for another 25 seconds until voicemail kicks in and answers the call at the 30 second mark. I get a voicemail message containing nothing but a dial tone (there is a problem with using tone_detect to avoid this, but that is a topic for another post). 2. First remote party calls, waits for 15 seconds, then hangs up. Second remote party calls 10 seconds later. 5 seconds after the second call starts, voicemail kicks in and answers, believing that there has only been one caller and that the phone has been ringing for 30 seconds. Now I accept that to some extent these kinds of problems are unavoidable without the assistance of something like ROIC. There is no immediate way to tell the difference between a gap between two RINGs and the remote caller having hung up. The difference only becomes obvious at the time when the next RING ought to have started (I suppose a RING that was cut short could also be detected, but since the gaps are longer than the RINGs it is more likely that the hangup will occur during a gap). Likewise, if a second caller comes along just perfectly starting at the instant that the next RING was due for the first caller then I can't see how FreeTDM can avoid being tricked. However, at present, FreeTDM totally ignores FTDM_OOB_RING_STOP events, and only seems to take notice of the first FTDM_OOB_RING_START event (well it counts rings, but doesn't seem to notice that the count has stopped increasing, at least not until more than 25 seconds after the ringing has stopped). This creates an unnecessarily long window of opportunity for problems. Given the Australian ring cadence (0.4s RING, 0.2s gap, 0.4s RING, 2s gap) it is possible to detect the caller having hung up within at most 2 seconds - simply reset a timer on each FTDM_OOB_RING_START and FTDM_OOB_RING_STOP event, and reset the line state if the timer gets over 2 seconds. I have implemented this (rather awkwardly), and it works well for me, but my timeout is hard-coded and therefore it would not work with a US cadence (2s RING, 4s gap). What I would ideally like to see is for FreeTDM to support RING cadence specification (both for detection on the FXO lines and for distinctive ring on the FXS lines) but that would be a big job and way out of my league. Failing that, there would need to be some config setting to control how long to wait for a RING_START/RING_STOP event before concluding the caller had hung up. Which config file would be the most logical place? freetdm.conf just in case different channels need different timeouts? Or is there some better approach to the whole problem? [I am a little surprised there has not been more complaint about these issues. If nobody else suffers from the same problems then perhaps I just have something misconfigured??]

1 0

Setting channel variables in FIFO recording template
by Peter Steinbach 05 May '12

05 May '12

There was already a thread for this in 2010 ("Setting channel variables for FIFO-originated calls" http://freeswitch-users.2379917.n2.nabble.com/Setting-channel-variables-for…) but my requirements are a bit different so I am asking this again: What am I going to do? I want to record a file in a fifo (agent is not a consumer) with the agent's id and caller_id_number and date. * if I set fifo_record_template in the dialplan, I do not have the agent's id available, because this (external) id shall not be the phone number and at the time of evaluation of the dialplan we cannot predict the agent's numer and it's id and the call date * If we enter the caller_id-number in the fifo config we can set the agent's number and id but we cannot set the caller_id_number and the date as Freeswith will not accept it 2012-05-05 23:03:53.517552 [CRIT] switch_channel.c:1183 Invalid data (${fifo_record_template} contains a variable) My current template is as follows: fifo_record_template=/usr/local/freeswitch/recordings/PROJ1/20120505/${caller_caller_id_number}__12345__202__20120505_224740__Sales.wav How can I overcome this problem? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

1 0

FreeSWITCH 1.2 and packaging update
by Ken Rice 05 May '12

05 May '12

As many of you know we have been working hot and heavy on getting 1.2 ready to drop. In the next 24 hours we will be pushing some further changes into tree that will affect a small number of users. These Changes are leading up to the 1.2 release and specifically address Packaging for Debian (and Debian based) system. If you arent currently using the debian package building bits of the FreeSWITCH source tree, you will not be affected. However if you are expect to see some big changes... These changes are to move us more inline with the Centos packages and more inline with what is required to get FreeSWITCH in the normal Debian repositories. Also, do we have a brave soul that might want to help with the FreeSWITCH Change log? Contact me off list or via IRC (SwK in #freeswitch on irc.freenode.net) if this is something you can help with. K

2 1

Re ceiving only RTP packets
by rajprithiv89 05 May '12

05 May '12

Hi, we are using csipsimple android voip application which uses Freeswitch as SIP/Media server. We established voip calls between two android clients(A, B) by enabling SRTP in both the sides. Now the problem observed here is, when we captured the packet flow in wireshark, the client which initiates the call(A) is sending SRTP packets and the client which receives the call sends only RTP packets and the FreeSwitch sends SRTP packets to both the clients. We have tested FreeSwitch with other application, there also we ended up in the same problem(one sided SRTP packets flow). Also we have tested the same CSipSimple application with Asterisk server. It works fine by sending SRTP packets from both the clients. When i captured the Wireshark packets SDP contains: >From A to Server: m= audio 4006 RTP/SAVP 98 97 99 >From Server to B m = audio 31562 RTP/SAVP 3 98 99 9 m = audio 31562 RTP/AVP 3 98 99 9 (sends two media line, one for SRTP and one for RTP) >From B to Server(Negotiation done) m = audio 4014 RTP/SAVP 3 96 m = audio 0 RTP/AVP 3 98 99 9 (This is to inform, B is not going to use RTP by putting the port number as 0) >From Server to A(Negotiation done) m = audio 32338 RTP/SAVP 3 96 But in the end, A is sending SRTP packets Server is sending SRTP packets B is sending only RTP packets (But in the negotiation RTP/AVP port was 0) -- View this message in context: http://old.nabble.com/Receiving-only-RTP-packets-tp33758451p33758451.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

3 2

gsmopen error message
by Yihui Li 05 May '12

05 May '12

Hi, I am excited by new mod_gsmopen with dongle and trying it with my E1550. The function is OK. But I got some error message when sending SMS Quote freeswitch@ubuntu> chat SMS|gsm01|07575565692|test 2012-05-04 23:40:11.102053 [ERR] gsmopen_protocol.cpp:574 rev 2431e0f|de019ab[(nil)|37 ][ERRORA 574 ][gsm01 ][-1, 0, 0] TIMEOUT=1100 2012-05-04 23:40:11.182283 [ERR] gsmopen_protocol.cpp:574 rev 2431e0f|de019ab[(nil)|37 ][ERRORA 574 ][gsm01 ][-1, 0, 0] TIMEOUT=5500 Sent Unquote It is under the default configuration. Does anyone know the meaning and how to avoid it? Thanks Regards Eric

3 3

Park and Page use case
by Ian McMaster 04 May '12

04 May '12

As this is my first question, please let me first praise the FreeSWITCH team for a fantastic product and all the efforts of the community. The two books really helped convert all my Asterisk AGI / AMI java code to FreeSWITCH. All IVR play/record/dtmf, etc is working perfectly with FreeSwitch now (and the IVR application is quite large). I came to FreeSwitch after limitations/frustrations with Asterisk. Happy to be aboard, and hope to contribute in time. Use case: Controlling an IVR caller using the event socket; caller chooses an option which will park the caller, and page out to a group of phones for someone to pick up the parked caller. If the park is answered the script exits, or if the park times out then an alternate action is taken. The valet_park command is working fine, and I can decipher through the ChannelExecuteComplete event whether the park was retrieved or not. I am struggling a bit with the paging to multiple polycom phones. - using something like 'originate {sip_auto_answer=true}user/1001,user/1002,... &playback(announce_page.PCMU) has limited success - my understanding is that the first phone to answer 'wins' and the other phones do not receive the playback. - I have better success with conference (a) Invite phones into the conference: conference page_group dial {sip_auto_answer=true}user/1001 ... etc (b) Play file: conference page_group play announce_page.PCMU (although I am having a codec problem here!) (c) End the conference: conference page_group hup all While the conference by nature has certain pre-stocked files (i.e. you are the first caller, and music on hold, etc), I believe I can silence them all with a custom conference. My only show stopper is that if a page goes out to a phone with two line buttons, and is already in use (on one line), then things go bad. The incoming conference request is knocking the existing call on hold, the page is playing out, and the user must resume the call with a soft button. Questions: 1) Is there a better approach to paging multiple phones that I have yet to try? 2) Is there a way to avoid paging a phone that already has a caller on it? 3) Is there conference variable to set the codec to allow the PCMU file to play cleanly (it does play cleanly using originate without setting any explicit codec). Thank you in advance, Ian.

2 1

Meaning of originate_timeout
by Malay Thakershi 04 May '12

04 May '12

Hello, When doing an originate, I set originate_timeout=45 Does it mean the called phone will ring for maximum 45 seconds? Thanks. Malay

2 1

Receiving "early media" during call
by Pete Kelly 04 May '12

04 May '12

Hi I have a scenario where there is an established call (originated from a freeswitch bridge to a downstream UAC) The downstream UA then initiates a re-invite and freeswitch accepts and keeps sending the 200OK with SDP as normal. However the downstream UA is also sending a 183 with sdp to freeswitch at this time, and it looks like freeswitch is ignoring this media. Is it possible to get freeswitch to accept this 183 media during a reinvite transaction? Pete

2 1

Todays Git update has errors
by Mario G 04 May '12

04 May '12

FYI - Been updating fine until today. Got the message below and update stopped. making all mod_event_socket Compiling /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c... quiet_libtool: compile: gcc -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -g -O2 -pipe -no-cpp-precomp -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c -fno-common -DPIC -o .libs/mod_event_socket.o cc1: warnings being treated as errors /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c: In function 'read_packet': /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:1360: warning: format '%lld' expects type 'long long int', but argument 9 has type 'time_t' make[6]: *** [mod_event_socket.lo] Error 1 make[5]: *** [all] Error 1 make[4]: *** [mod_event_socket-all] Error 1 make[3]: *** [all-recursive] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2

3 5