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park_after_bridge does not work if far end does 180 Ringing
by Ali Pey 17 Mar '15

17 Mar '15

Hello, I see a different behaviour for 183 Ringing vs 180 Ringing and a 408 Request Timeout response message with regards to park_after_bridge: 1) I have "park_after_bridge=true". 2) Do a bridge to a PSTN number (SIP Trunk) 3) After sometime freeswitch receives 408 Request Timeout message: a) if carrier had sent 183 Ringing (in band ring back tone) the a-leg is parked after the 408 timeout message. b) if carrier had sent 180 Ringing, the aleg is hungup after timeout (408 message) Is this a bug? Is there a work around? Thanks, Ali Pey

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delay before ringback at 183 to 180 transition
by Jeff Pyle 17 Mar '15

17 Mar '15

Hello, FreeSWITCH Version 1.5.15b+git~20150313T215013Z~9872c52c57~64bit (git 9872c52 2015-03-13 21:50:13Z 64bit) I have a situation where a gateway sends a 183 with media, then a 180. Following Anthony's advice from FS-3859 I have the following: <extension name="sendit"> <condition field="destination_number" expression="(.*)"> <action application="set" data="ringback=/usr/local/freeswitch/sounds/ringtester.wav"/> <action application="bridge" data="{bridge_early_media=true}sofia/public/$1@${host2}"/> </condition> </extension> On the a-leg I hear media from the 183 until host2 stops it and sends the 180. Then there's a 4.8 second delay of no RTP from FS before I hear my ringtester.wav file. This occurs with a tone stream as well. The last event on the console is "Callstate Change EARLY -> RINGING" when the 180 comes in, then "nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering" when the 180 arrives. Then 4.8s of nothing. Then FS resumes the RTP with the ringback file. There is nothing new displayed when the RTP resumes with the ringback file. Is this a bug, or a misconfiguration? - Jeff

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Port changes in multiple 183s cause no audio after 200
by Jeff Pyle 17 Mar '15

17 Mar '15

Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit). The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with. The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example: - Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183). In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome! - Jeff

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read application in dialplan is notworking when I use bind_meta_app application ( FXS, TDM 400P with DAHDI )
by jack 17 Mar '15

17 Mar '15

To whom it may concern, I just hit the wall when I test the atended transfer. Here's my simple dialplan: test.xml: <extension name="DID-Call"> <condition field="destination_number" expression="^(382)$"> <action application="bind_meta_app" data="1 b s execute_extension::att_xfer XML features"/> <action application="bridge" data="freetdm/myFXS/1"/> </condition> </extension> features.xml: <extension name="att_xfer"> <condition field="destination_number" expression="^att_xfer$"> <action application="answer"/> <action application="read" data="4 4 'tone_stream://%(5000,0,350,440)' digits 5000 #"/> <action application="att_xfer" data="user/${digits}"/> </condition> </extension> And the test environment configuration is like this: As you can see, when you dial number "382" from phone 282, the Diguim telephony card will get ring signal at once. Then the freeSWITCH server starts a thread to handle dialplan. While in the execute-state, it will get in the test.xml and match the condition, so FreeSwitch will help bridging two channels together (freetdm/myFXO and freetdm/myFXS/1). After bridging, you'll hear the ring from phone 4001 and pick up the phone, then you can dial "*1" to activate the action and transfer to call featurs.xml immediately. You're now in features.xml and matched the exact condition "att_xfer", so you will hear the generated dial tone by tone_stream and need to dial 4 digits in the moment. The BIG problem is, I can't get any DTMF tone while hitting the digit keys on the phone 4001. But when I change the dialplan like this: test.xml <extension name="DID-Call"> <condition field="destination_number" expression="^(382)$"> <action application="bind_meta_app" data="1 b s execute_extension::att_xfer XML features"/> <action application="bridge" data="user/1001"/> </condition> </extension> The "read" works just fine and really do the addended transfer as expect. And I do some tests to make sure the read function work well in normal context, which means, I discard the "bind_meta_app" application, just use "read" when phone has been answered as follow: test.xml <extension name="DID-Call"> <condition field="destination_number" expression="^(382)$"> <action application="answer"/> <action application="read" data="4 4 'tone_stream://%(5000,0,350,440)' digits 5000 #"/> <action application="log" data="The number you dialed is : ${digits}"/> </condition> </extension> and it works well when I dialed 382 and reach out the freeswitch server, I could see the DTMF tone show on console. After I dialed "4001", the "log" application shows "The number you dialed is : 4001", just likes I expected. So, why would I get involved with situation like this? I was checked the mod_dptools.c and nothing is found in there, since the arguments are not the trouble maker. Maybe the critical part came from FreeTDM( samgoma ) Module, but I gotta figure out which direction of thought is the most probable. If you guys could give me some advice I'll be very thankful, thanks in advance! Kind Regard, Jack Huang

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freeswitch sip trace to separate file
by mbo 17 Mar '15

17 Mar '15

Hello, is it possible to configure freeswitch to write the SIP Trace to a separate file, preferable different files per sip profile? If no, is there already a feature request for that, otherwise I create one? Thanks and Regards Markus

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FS voicemail notification to mobile SMS?
by Terry Barnum 17 Mar '15

17 Mar '15

When someone leaves a voicemail at the house, FS + FusionPBX successfully emails the Google transcription along with the audio attachment of the voicemail. It's working well. High Spousal Approval Factor. Now I would like to send a SMS text message notification via cellular to our mobile phones when a voicemail is left. Possible? I don't necessarily need the audio of the voicemail just a notification with the caller id in the SMS. (Though if it's easy I'll try it.) I've looked at the mod_sms wiki docs but all the examples appear to be between FS clients. How to send out to a cellular mobile number? Anyone have working examples they can share? Thanks, -Terry

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DTMF (Mod-Call Center)
by Ali Jibran 16 Mar '15

16 Mar '15

I wanted to know if there is some way we could find out what DTMF was pressed when agent picks up the the call. Like if there is a queue say support@default on the same Extension but differnet numbers..so how can I know what DTMF was pressed to enter it?

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FreeSWITCH Week in Review (Master Branch) March 7th-14th
by Ken Rice 16 Mar '15

16 Mar '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1CmA0mR FreeSWITCH Week in Review (Master Branch) March 7th-14th Hello, again. This passed week in the FreeSWITCH master branch we had 13 commits. The features for this week include: the addition of Australian ringback, a minimal configuration for configuring FreeSWITCH from scratch, and the filter feature ported from mod_event_socket to mod_erlang_event . Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7354 Filter feature ported from mod_event_socket to mod_erlang_event FS-7362 Add minimal configuration for configuring FreeSWITCH from scratch FS-7368 Added Australian ringback The following bugs were squashed: FS-7355 Fix rpl_realloc symbol missing link error that can occur when using clang. FS-7300 Handle all MRCP completion causes in SPEECH-COMPLETE event and validate load input grammar URLs

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Caller ID not logged in CDR properly
by Alexander Baldeck 16 Mar '15

16 Mar '15

Hello, we are using Freeswitch 1.5.12b+git~20140324T234505Z~4dd0a5848f~64bit interfacing it with Plivo and experience an issue where our call detail records list the destination_number twice when someone picks up a call like so: "Outbound Call","99901637777777","99901637777777","default","2015-02-28 13:15:00","2015-02-28 13:15:08","2015-02-28 13:16:32","92","84","NORMAL_CLEARING","24fdbb20-32a3-11e4-b883-e58f73531fe4","","","PCMA","PCMA" If the call isn't picked up upon though, everything is alright and caller_id_number is logged alongside destination_number like so: "","0123456789","99901637777777","default","2015-02-28 13:11:00","","2015-02-28 13:12:00","60","0","NO_ANSWER","95b050f4-32a2-11e4-b87f-e58f73531fe4","","","","" The used template in cdr_csv.xml looks like this and I made sure it is being used: <template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template> I'm at a loss what's going on here, has anyone got insight on where to look? Best, Alex

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Transfer back to origin on failure
by Paul Atreides 16 Mar '15

16 Mar '15

When I do a blind transfer then I want freeswitch to call back the origin who initiated the call. But I am not able the capture the transfer event? <condition field="destination_number" expression="^att_xfer$"> <condition field="destination_number" expression="^is_transfer$"> They seem do be ignored by the dialplan. Is there a list what kind of values destionation_number can have besides the called numbers? I am doing the transfer with a grandstream gxp2140

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