- FreeSWITCH-users - FreeSWITCH Mailing Lists

context of transfer call
by Bruce Marriner 29 Apr '19

29 Apr '19

I'm having the issue that when calls come in from the outside they hit the public context which sends the call to a phone, and then user at the phone tries to transfer them to another extension, then call is dropped. That's because, the transfer gets put in the "public" dialplan where the call came in from. I've read that other's have had this issue, and surely this is a normal use-case to transfer inbound calls, so I'm hoping this is a solved issue :) I read some things about using force_transfer_context but I can't seem to figure out how to use it. I though maybe it was a variable so I tried adding it as a variable in the user directory but that didn't help :( Can someone help explain how to address this issue so inbound calls can be transferred? Thanks. :) Ideally, I'd like FreeSWITCH to use the context of the phone/user that's doing the transfer.

2 2

Sending periodical audio announcement to callee
by Arik Halperin 29 Apr '19

29 Apr '19

Hello, I would like to play an audio recording to the callee of a call every few seconds. What is the best way to do this? Thanks, Arik Halperin

2 1

Record_ession and SIP INFO record: on
by Allan Kristensen 28 Apr '19

28 Apr '19

Hello, I'm having some issues getting dialplan command "record_session" and SIP INFO messages to work together. If recording is started using record_session in the dialplan, at the beginning of the call and I then send a SIP INFO message with "record: on" i get the response "Already recording", which is as expected. To avoid this situation, we then send the recording state in response header, so SIP client knows if recording is already started at the begining of the call. Now if I try to stop the recording using SIP INFO message, I get the error message "Nothing to stop"? I'm guessing it has something to do with recording bug being in a different session of the call. Is there anyway to adjust this behaviour? (One could argue that just because I send a SIP INFO to stop the recording does mean that it should be allowed) Thanks,

2 1

originate with SIT tones
by John Burke 26 Apr '19

26 Apr '19

Hello: Does the originate command support monitor_early_media_fail like the following: originate {ignore_early_media=true,monitor_early_media_fail=user_busy:2:480+620!destination_out_of_order:2:1776.7}sofia/gateway/test/1000 All examples online that I have seen are for the bridge application, but I am not having much luck with originate. I made several test calls on FS1.8.5 with 480/620 tones in the 183, but originate progresses normally without firing an event or hanging up the call (not sure which one is expected). Anyone have any examples, configs, etc? Thanks, John

1 0

No Video in conference video mux mode with VP8 but H264 works fine
by Faisal Hanif 26 Apr '19

26 Apr '19

I have a conference with video-mode mux and I have proper mixed video with H264 but with VP8 only empty canvas sent by FreeSWITCH while video-mode passthrough send proper video with VP8. Any idea what's wrong or missing here? Regards, Faisal

4 6

VoiceMail to Email has blank WAV file
by Sean Devoy 26 Apr '19

26 Apr '19

Hi All, I have been trying to get VoiceMail to email working in the latest production version (1.8) on Ubuntu. I am using sendmail and I have gotten as far as sending the email with an attachment. The attachment is a wav file of the correct length but has no audio. It is for example 5 seconds of silence. In the voicemail folder for this extension is a wav file with audio just fine. The logfile says: 2019-04-26 04:01:46.842091 [DEBUG] switch_utils.c:1210 Emailed file [/tmp/mail.155625130604d1] to [sdevoy(a)bizfocused.com] 2019-04-26 04:01:46.842091 [DEBUG] mod_voicemail.c:3042 Sending message to sdevoy(a)bizfocused.com<mailto:sdevoy@bizfocused.com> I have checked to protection on the /tmp folder. Any ideas what is wrong? Thanks, Sean

1 1

ESL API command to get the status of Freeswitch
by Ramesh Kandasamy 25 Apr '19

25 Apr '19

Hi everyone, I am looking for a way to get the status of a Freeswitch from a remote server through ESL API command. I tried using SendRecv and Api methods of ESLconnection class to send the 'status' command. But I don't receive the response. However, after further reading through the confluence, I realized that it is probably because, 'status' command output doesn't include with a header named "content-type" that has a value of "api/response" or "command/reply" and both SendRecv and Api methods look for this. So, the question is what other ESL methods shall be used to get the status of freeswitch? Thanks Ramesh

3 4

Bridge question
by Sean Devoy 25 Apr '19

25 Apr '19

Hi, My customer wants to ring the local extension, then 15 seconds later connect their cell phone. I have the leg_delay_start working fine. What I don't know is how to pass the internal Caller ID to one and the external caller id to the other? Thanks, Sean

2 3

how to connect FS to openSIPS
by Mundkowsky, Robert 25 Apr '19

25 Apr '19

I have openSIPS configured as a load balancer behind FreeSWITCH. I setup a sip profile for openSIPS in /etc/freeswitch/sip_profiles/internal/engagement.xml <include> <gateway name="engagement"> <param name="realm" value="XXXX"/> <param name="username" value="1000"/> <param name="from-user" value="1000"/> <param name="from-domain" value="freeswitch.internal.org:5060"/> <param name="password" value="1000"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="ping" value="25"/> </gateway> </include> Then setup a diaplan that should call openSIPS: ... <action application="conference_set_auto_outcall" data="{sip_from_display=${client_session_id}}sofia/gateway/engagement/${as_ext}${destination_number}"/> ... When call is made, I don't see any errors in the FreeSWITCH log and nothing is seen in the openSIPS log. I suspect I have an error in the sip profile. ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

1 1

Caller ID NAme switches to "mod_sofia"
by Sean Devoy 23 Apr '19

23 Apr '19

Hi, My customer has Aastra 67xx phones. I have two problems relating to caller id. 1 - The setup on the phone appears to take president over my setup for that extension in the XML (for caller id and caller name). 2 - While the correct number and extension appear from interoffice phones, as soon as it gets answered the display changes to "mod_sofia". Any advise will be greatly appreciated. Thanks, Sean

1 0