February 2022 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Python Development
by Joli Martinez 21 Feb '22

21 Feb '22

Hello, I have written some scripts in Lua for freeswitch, but I would like to switch over to Python since I am more familiar with the language. I am not sure how to install the freeswitch module. I would be running Debian 10 or 11. I see that there are examples online but they all require importing freeswitch. When I run pip, all the available pip packages with the word Freeswitch is ESL or Eventsocket. When researching the ESL setup is different. So I am sort of confused. Could someone point me in the right direction as to how to install the correct packages and a sample "Hello World" script. I should be able to take it from there.

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Limit concurrent calls for inbound + outbound on a specific DID number
by Rasheed Kalapurackal 21 Feb '22

21 Feb '22

Hello , Is it possible to limit the total number of inbound + outbound calls made or received on a specific DID number to 10 ? Thanks Rasheed

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Media timeout
by Shaun Stokes 21 Feb '22

21 Feb '22

Hi All, I'm not sure if others have had similar experiences but for us the media_timeout variable does not work as expected. * media_timeout on a call that supports video but with-out video will fail. * media_timeout_audio works in some instances, in others the timeout period is ignored so the call will timeout almost immediately after RTP stops. * media_hold_timeout_audio doesn't seem to work at all, calls that are on hold never timeout. Why is the SIP profile parameter 'rtp-timeout-sec' depreciated? It's much simpler to apply this per SIP profile than it is per call. Thanks, Shaun

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voicemail from domain
by Kim Culhan 21 Feb '22

21 Feb '22

Running FreeSWITCH Version 1.10.3-release~64bit (-release 64bit) Emails sent with voicemail messages have a from line: extension@ip_address_of_external interface. In voicemail.conf.xml have: <param name="email-from" value="${voicemail_account}(a)mydomain.com"/> This did not change it, still has @external_ip_address Any help is greatly appreciated. thanks -kim

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Why did the user phone send a BYE.
by Sean Devoy 18 Feb '22

18 Feb '22

I have reviewed my pcap file MANY times. I have a FAILED call and a GOOD call from the same extension. I will attempt you list the signaling below. One oddity that jumps out at me, there are no SIP 183 packets in the call that WORKED. It is clear that the user is hanging up, but they state the call rings 4 times, then goes to busy. The tshark command I am using is: tshark -f "port 5060" -i any -w capturefile.pcap How can I include RTP messages? Will that let me see if there is early media, then busy or failure? Call that failed signaling: U is the User extension behind NAT. FS is the server with public IP. GW is gateway with public IP (Voip Innovations). U-> FS INVITE FS->U AUTH REQ U->FS Request: ACK U->FS INVITE FS->U Trying FS->GW INVITE GW->FS TRYING U->FS Status 200 GW->FS Status: 183 Session Progress FS->U Status: 183 Session Progress 1/10 Second Later GW->FS Status: 183 Session Progress 21 Seconds Later GW->FS Status: 183 Session Progress GW->FS Status: 200 FS->GW Request: ACK FS->U Status 200 U->FS Request ACK FS->U Request: UPDATE U->FS Status 200 3 Seconds Later - 26 seconds total U->FS BYE FS->U Status: 200 FS->GW BYE GW->FS Status: 200 Regards, Sean Devoy VP Operations and Development Business Focused Internet Systems, Inc.

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Single RTP port to handle all RTP streams
by Mahmood Alkhalil 18 Feb '22

18 Feb '22

Hi all, Is it possible to configure FS to use a single port for all RTP streams? I face tons of issue with our ISP blocking most of the ports you might think of, so is it possible to do such with FS or any other software i can use in front of freeswitch?

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mod_sms: use different context
by António Silva 16 Feb '22

16 Feb '22

Hi, Is it possible to set a different context in chatplan.xml? Or when receiving a chat in the default context send it to a different context like transfer app? I can set the context in Sofia profile, but is not possible to set it per user… The following didn't work: <extension name="default_catch_all"> <condition field="to" expression="^(.*)$"> <action application="transfer" data="${to} XML outgoing"/> </condition> </extension> Got the following error: Chatplan: 6 parsing [default->tapi] continue=false Chatplan: 6(a)lab2.local Regex (FAIL) [tapi] to(6(a)lab2.local) =~ /^tapi.*$/ break=on-false Chatplan: 6 parsing [default->tel message sensor] continue=false Chatplan: 6(a)lab2.local Regex (FAIL) [tel message sensor] to(6(a)lab2.local) =~ /^tel_message_sensor/ break=on-false Chatplan: 6 parsing [default->default_catch_all] continue=false Chatplan: 6(a)lab2.local Regex (PASS) [default_catch_all] to(6(a)lab2.local) =~ /^(.*)$/ break=on-false Chatplan: 6(a)lab2.local Action transfer(${to} XML outgoing) 2022-02-15 09:45:14.759408 83.57% [ERR] switch_loadable_module.c:1000 Invalid chat application interface [transfer]! 2022-02-15 09:45:14.759408 83.57% [DEBUG] sofia_presence.c:225 Can't find registered user 6(a)lab2.local -- Saludos / Regards / Cumprimentos António Silva

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How to add text outside P-Asserted-Identity
by Adiseshu Channasamudhram 15 Feb '22

15 Feb '22

Hello Team Am trying to add a text tag to the P-Asserted-Identity as below to make my app where the INVITE is coming from <action application="export" data="sip_cid_type=none"/> <action application="export" data="sip_h_P-Asserted-Identity='TextTag';sip:${sip_P-Asserted-Identity}@${distributor(fidelity-site_p)};user=phone"/> When I do the above, the TextTag is getting included within the P-Asserted-Identity and hence when the call is transferred, the carrier will reject saying it is Invalid P-Asserted-Identity So - how can I add a TextTag to P-Asserted-Identity but outside ..... Currently it is adding as, P-Asserted-Identity: <'TextTag';sip:+13333333333;oli=00@10.6.3.7:5060;user=phone> How can i make the above as P-Asserted-Identity: "TextTag" <sip:+13333333333;oli=00@10.6.3.7:5060;user=phone> Thanks a lot in advance Regards Adi

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Transfer fails when P-Asserted-Identity is modified
by Adiseshu Channasamudhram 15 Feb '22

15 Feb '22

Hello Team I have a requirement to add a tag and also copy the P-Asserted-Identity from the incoming INVITE to the outgoing INVITE I tried to use the below config in dialplan and this did change the P-Asserted-Identity value as required but then when the call is made to transfer , it fails <condition field="network_addr" expression="^(19\.1\.24\.109|19\.1\.27\.109)$"> <action application="export" data="sip_cid_type=none"/> <action application="export" data="sip_h_P-Asserted-Identity='CarrierName';sip:${sip_P-Asserted-Identity}@${distributor(fidelity-site_p)};user=phone"/> </condition> Can someone please check and let me know if the above method is correct or what is the best way to achieve this Regards Adi

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Prevent Freeswitch from copying RPID in 200 OK response to A-leg
by Marcin Leśniewski 15 Feb '22

15 Feb '22

Hello everyone, I have a problem with UAC sending RPID in 200 OK response which is then copied by Freeswitch to A-leg. All methods I've tried so far allowed for removal or modification of RPID header in outgoing INVITE on B-leg, but when B-leg sends 200 OK response, it contains RPID header which is copied and passed down to A-leg, messing up with caller id displayed on the phone. Is there a way to stop this behaviour? Below is a diagram of an example call. UAC A FS UAC B | INVITE | | | contains RPID | | |----------------------------------->| INVITE | | | no RPID (correct) | | | ------------------------------->| | | | | | 200 OK | | | contains RPID | | 200 OK |<--------------------------------| | contains RPID, copied from B-leg | | | I want this RPID removed | | |<-----------------------------------| | Cheers, Marcin

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