- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to copy from header
by Adiseshu Channasamudhram 23 Feb '22

23 Feb '22

Hello Team I have a requirement to copy from header from the incoming INVITE to the outgoing INVITE right now, below is the from header from incomign invite From: "123-456-7788" <sip:+11234567788;cpc=ordinary@12.18.10.15:5060;user=phone;isup-oli=0>;tag=2bf345ee but the outgoing INVITE has the from header as below From: "123-456-7788" <sip:+11234567788@18.1.8.19>;tag=yaXFUFjD666ym How can i copy the cpc and isup-oli from the incoming from header to the outgoing header? Thanks in advance Regards Adi

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Read Header
by Alexander Perkins 23 Feb '22

23 Feb '22

Hi All. I have a custom header that will be passed down to be by our provider and I would like to read that header. How can I do that? It is one of those X- headers. Thank you, Alex

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Re: [Freeswitch-users] voicemail from domain
by Bote Man 23 Feb '22

23 Feb '22

Sean, what is the difference between the user's extension and the voicemail extension? They should be the same, no? I found a plethora of variables and parameters that you can set in: https://freeswitch.org/confluence/display/FREESWITCH/mod_voicemail Hope this helps. John Boteler Bote Communications -----Original Message----- From: FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> On Behalf Of Sean Devoy via FreeSWITCH-users Sent: Sunday, 20 February, 2022 22:50 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] voicemail from domain Hi, I finally found that setting these values in the directory file, specifically the voicemail directory for the user works. Here is an example: <param name="vm-password" value="1234"/> <param name="vm-message-ext" value="wav"/> <param name="vm-delete-file" value="false"/> <param name="vm-attach-file" value="true"/> <param name="vm-mailto" value=beth(a)somedomain.com/> <param name="vm-mailfrom" value=voicemail(a)somedomain.com/> <param name="vm-email-all-messages" value="true"/> I first made the mistake of adding these to the user’s extension, but it must be that user’s voicemail extension. HTH, Sean

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Unable to establish end-to-end OPUS communication with installation freeswitch configuration
by MG PX 23 Feb '22

23 Feb '22

Freeswitch is at ip address 192.168.0.27 My softphone #1 is registered with freeswitch as user 1005. It is at ip address 192.168.0.30 My softphone #2 is registered with freeswitch as user 1006. It is at ip address 192.168.0.16 1005 calls 1006 using OPUS codec. The call does not succeed because the media stream (audio only in my case) initially established between FS and the caller 1005 is interrupted after FS has setup the communication with callee 1006 and sent the SIP OK to caller 1005. All other media streams are correctly established. I got this behaviour only with OPUS codec. It works correctly with G722 codec for instance. I tried with freeswitch 1.8.7 under windows and freeswitch 1.10.7 under linux and got the same result. I tried different freeswitch settings with the same result. Can someone help me to solve this problem? I don't know where to look anymore. I attached a wireshark capture. It shows that FS is not sending media packets anymore to 1005 (192.168.0.3) after it has sent the SIP OK (packet #2360). I also attached freeswitch log.

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Re: [Freeswitch-users] PCAP help and many call failures PLEASE HELP
by mayamatakeshi 22 Feb '22

22 Feb '22

On Wed, Feb 23, 2022 at 2:20 AM Sean Devoy via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Sean Devoy <sdevoy(a)bizfocused.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 22 Feb 2022 16:56:15 +0000 > Subject: PCAP help and many call failures PLEASE HELP > Hi, > This is a very serious problem for my users AT this site. PLEASE help. > > I suspect it is a NAT issue reaching the extension. I have tried <variable > name="sip-force-contact" value="NDLB-connectile-dysfunction"/> with no > change. > I have tried TCP instead of UDP. > > Details of the problem: > User extension initiates a call (INVITE after auth required). > User hears ringing for approximately 20 seconds, then a busy signal. Today > a user called her own cell phone and said it WAS ringing when the call > failed. > I see some RTP packets between 183 session progress from FS to extension > and CANCEL, but can't tell what extension hey are to/from. > I see a very single strange RTP from FS to itself: unknown RTP protocol > version 3. > The call CANCEL is initiated from FreeSwitch at 22.75 seconds after the > invite to FS. > > PCAP issue: > I am trying to trace down a problem with tshark. > The short version of my question is how do I tell which NAT extension RTP > packets are going to and why did FS cancel the call? > > The full explanation is: > I have identified the failed call based on the INVITE from the NAT > extension's IP and PORT number. I am able to follow the SIP packets is > OK. I see RTP packets to and from that IP address, but I cannot determine > what extension it is going to. > > I also see an ICMP from the FSserver to the extension's external IP aftern > the CANCEL. We block ICMP. Is that a problem? > > I have the full pcap file if someone could PLEASE help > Pcacp file: http://brianbunce.com/capturefile.pcap (11 MB) It is a > customer site, be kind. > The invite from the ext is packet# 2334. The invite to the GW is packet# > 2357. That will show you all IPs involved. > > I am not a security expert and I am not sure if it is wise to share capture files with unredacted information. Anyway, I took a look and I think what matters is to understand why FS decided to cancel the call. If you have the XML CDRs of the channels involved, they might give some clue about it. If you don't have them, please show the complete dialplan you are executing as there might be something there like call_timeout etc. Then even if with those things the cause of CANCEL cannot be found, one alternative if you are able to reproduce this problem at will or if the problem happens frequently enough would be to try to enable debug logging during low traffic period by doing: fs_cli -x "fsctl loglevel DEBUG" +OK log level: DEBUG [7] Then make a call that recreates the problem and switch back to the original log level (usually, ERR): fs_cli -x "fsctl loglevel ERR" +OK log level: ERR [3] Then with the logs, you should be able to understand why FS is deciding to cancel the call.

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PCAP help and many call failures PLEASE HELP
by Sean Devoy 22 Feb '22

22 Feb '22

Hi, This is a very serious problem for my users AT this site. PLEASE help. I suspect it is a NAT issue reaching the extension. I have tried <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> with no change. I have tried TCP instead of UDP. Details of the problem: User extension initiates a call (INVITE after auth required). User hears ringing for approximately 20 seconds, then a busy signal. Today a user called her own cell phone and said it WAS ringing when the call failed. I see some RTP packets between 183 session progress from FS to extension and CANCEL, but can't tell what extension hey are to/from. I see a very single strange RTP from FS to itself: unknown RTP protocol version 3. The call CANCEL is initiated from FreeSwitch at 22.75 seconds after the invite to FS. PCAP issue: I am trying to trace down a problem with tshark. The short version of my question is how do I tell which NAT extension RTP packets are going to and why did FS cancel the call? The full explanation is: I have identified the failed call based on the INVITE from the NAT extension's IP and PORT number. I am able to follow the SIP packets is OK. I see RTP packets to and from that IP address, but I cannot determine what extension it is going to. I also see an ICMP from the FSserver to the extension's external IP aftern the CANCEL. We block ICMP. Is that a problem? I have the full pcap file if someone could PLEASE help Pcacp file: http://brianbunce.com/capturefile.pcap (11 MB) It is a customer site, be kind. The invite from the ext is packet# 2334. The invite to the GW is packet# 2357. That will show you all IPs involved. Thanks for any help, Sean

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freeswitch stuck at Push codec L16:100
by אריה קלטר 21 Feb '22

21 Feb '22

Hello group I have problem with one of our servers, that sometimes it starting to stuck, on all new calls the last log line is "push codec L16:100" and nothing helps to end the call, no response to bye from the calling server, no response to sip info and even no response to hupall, the only thing that solves the problem is restart the freeswitch itself. It happened two times. Any idea what is going on? Example from one of the calls? fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% [NOTICE] mod_dptools.c:1419 Channel [sofia/external/<NUMBER>@<IP>] has been answered fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% [DEBUG] switch_channel.c:3950 (sofia/external/ <NUMBER>@<IP> ) Callstate Change EARLY -> ACTIVE fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% [DEBUG] sofia.c:7499 Channel sofia/external/ <NUMBER>@<IP> entering state [completed][200] fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% [DEBUG] switch_ivr.c:632 sofia/external/ <NUMBER>@<IP> Command Execute [depth=1] conference(conf10@default+flags{}) fe4849de-cb96-4a4a-ad02-dc7485cc87ee EXECUTE [depth=1] sofia/external/ <NUMBER>@<IP> conference(conf10@default+flags{}) fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% [DEBUG] sofia.c:7499 Channel sofia/external/ <NUMBER>@<IP> entering state [ready][200] fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% [DEBUG] conference_member.c:1794 Raw Codec Activation Success L16@8000hz 1 channel 20ms fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 08:42:36.832268 92.10% [DEBUG] conference_member.c:1841 Raw Codec Activation Success L16@8000hz 1 channel 20ms and the next line is fe4849de-cb96-4a4a-ad02-dc7485cc87ee 2022-02-14 18:15:31.312291 95.10% [NOTICE] switch_core_session.c:407 Hangup sofia/external/ <NUMBER>@<IP> [CS_EXECUTE] [MANAGER_REQUEST] because i did hupall, but the call is still there until restart of the freeswitch itself, even after the hupall. Regards, Arye

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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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