- FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP Presence errors
by Bob McCarthy 10 Apr '18

10 Apr '18

Getting a lot of these errors when running in debug presence. Any idea of what's wrong ? 2018-04-09 18:29:04.906215 [ERR] sofia_presence.c:2308 SEND PRES NOTIFY: file[sofia_presence.c] func[broadsoft_sla_notify_callback] line[3380] profile[internal] via[] ip[192.168.57.101] port[5060] route[(null)] contact["Dispatch 1" <sip:SharedLineFour@192.168.57.101>] to[<sip:SharedLineFour@192.168.57.218:5060>;tag=rBf5t5vnTDXD;tag=rBf5t5vnTDXD;tag=rBf5t5vnTDXD] from["Dispatch 1" <sip:SharedLineFour@192.168.57.218>;tag=12E7BD9F-120FDB1E] url[sip:SharedLineFour@192.168.57.101] call_id[2cc83dc799e9fe8fa319dfc0f06948f2] expires_str[] event[call-info] ct[] pl[] call_info[<sip:192.168.57.218>;appearance-index=*;appearance-state=idle] exptime[55] 2018-04-09 18:29:05.486200 [ERR] sofia_presence.c:3775 DELTA 55 2018-04-09 18:29:05.486200 [ERR] sofia_presence.c:3897 check subs sql: select contact from sip_subscriptions where call_id='408b2bcd-50fcd4ce-f6bba5b3(a)192.168.57.103' and profile_name='internal' and hostname='CO-999-9' ["Dispatch 1" <sip:SharedLineTwo@192.168.57.103>] 2018-04-09 18:29:05.486200 [ERR] sofia_presence.c:3931 re-subscribe event call-info, sql: update sip_subscriptions set expires=1523320200, network_ip='192.168.57.103',network_port='5060',sip_user='SharedLineTwo',sip_host='192.168.57.218',full_via='SIP/2.0/UDP 192.168.57.103;branch=z9hG4bKc04f50ae65841513;rport=5060',full_to='<sip:SharedLineTwo@192.168.57.218:5060>;tag=5C0qZydXfunD;tag=5C0qZydXfunD;tag=5C0qZydXfunD',full_from='"Dispatch 1" <sip:SharedLineTwo@192.168.57.218>;tag=DCA332F6-E519577B',contact='"Dispatch 1" <sip:SharedLineTwo@192.168.57.103>' where call_id='408b2bcd-50fcd4ce-f6bba5b3(a)192.168.57.103' and profile_name='internal' and hostname='CO-999-9'

1 0

sip_presence
by Bob McCarthy 10 Apr '18

10 Apr '18

Is there any way to access the sip_presence db from the api ? Have documented a memory leak in presence FS-10808 and I would like to troubleshoot its root cause. Bob McCarthy

2 8

Re: [Freeswitch-users] How to get 200 Contact to use FQDN instead of IP
by E Chu 09 Apr '18

09 Apr '18

Hi Michael, The SIP client is an iOS app behind a NAT64 network interfacing to Freeswitch which is on an IPV4 network. In order for the 200 from FS to be properly acknowledged, the client needs to send the ACK to the host in the Contact field of the 200. However, if the Contact of the 200 contains the IPV4 address of FS, the client isn't able to send the ACK to the IPV4 address from the NAT64 network. So we're hoping that we can get FS to send its FQDN in the Contact field of the 200. Any suggestions appreciated. Thanks! ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 9 Apr 2018 12:17:21 -0400 > Subject: Re: [Freeswitch-users] How to get 200 Contact to use FQDN instead > of IP > why would you want to do this? > > On Apr 9, 2018, at 12:33 AM, E Chu <freeswitch-users(a)enlai.net> wrote: > > Hi all, > > Does anyone know how I can get FS to respond with a FQDN instead of an IP > address in the contact field of the 200? > > The domain is set to FQDN in vars.xml: > > [hostnames and IPs changed for public list privacy] > > <X-PRE-PROCESS cmd=“set” data="domain=fs-dev1.sipserver.com"/> > > i.e. but this is what FS is sending in the 200: > > Contact: <sip:123123123@1.2.3.4:5060;transport=tcp> > > I’d like it to be: > > Contact: <sip:123123123@fs-dev1.sipserver.com:5060;transport=tcp> > > This is the full 200 response from FS: > > SIP/2.0 200 OK > Via: SIP/2.0/TCP 192.168.0.1:52631 > ;rport=42810;branch=z9hG4bKPjwPkMUAMcou1XPnDrrWJH9GDwdlaPKXCF;alias;received=2.3.4.5 > From: "1020" <sip:1020@fs-dev1.sipserver.com > >;tag=h9ydSS-jO2dQ7xT8yxpVcLu-appl6Xtl > To: <sip:123123123@fs-dev1.sipserver.com>;tag=vXtN8yZgtHy0B > Call-ID: Y3tWlOhNz3f9bt-uHO-LjFjh2RQwQh9l > CSeq: 13167 INVITE > Contact: <sip:123123123@1.2.3.4:5060;transport=tcp> //HOW TO GET THIS TO > USE FQDN > User-Agent: FreeSWITCH-mod_sofia/1.6.16~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > Require: timer > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, > line-seize, call-info, sla, include-session-description, presence.winfo, > message-summary, refer > Session-Expires: 120;refresher=uac > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 248 > Remote-Party-ID: “123123123" <sip:123123123@fs-dev1.sipserver.com > >;party=calling;privacy=off;screen=no > > v=0 > o=FreeSWITCH 1522636701 1522636702 IN IP4 1.2.3.4 > s=FreeSWITCH > c=IN IP4 1.2.3.4 > t=0 0 > m=audio 18966 RTP/AVP 0 96 > a=rtpmap:0 PCMU/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=ptime:20 > a=rtcp:18967 IN IP4 1.2.3.4 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

1 0

caller redirect
by Paul Muaddib 09 Apr '18

09 Apr '18

Hi, one of our products has a service number which we purchase externally from a service provider. Now we don't want to change the numbers on our documents every time we change the service provider. Our company is located in Germany, the current service provider in the USA. Can I simply forward the incoming calls to the phone number in the USA with the "deflect" application without any costs for us? What about consumer protection? For example, the customer calls from Germany or Europe, but then has to make an international call with additional costs. Regards, Paul

1 0

How to get 200 Contact to use FQDN instead of IP
by E Chu 09 Apr '18

09 Apr '18

Hi all, Does anyone know how I can get FS to respond with a FQDN instead of an IP address in the contact field of the 200? The domain is set to FQDN in vars.xml: [hostnames and IPs changed for public list privacy] <X-PRE-PROCESS cmd=“set” data="domain=fs-dev1.sipserver.com"/> i.e. but this is what FS is sending in the 200: Contact: <sip:123123123@1.2.3.4:5060;transport=tcp> I’d like it to be: Contact: <sip:123123123@fs-dev1.sipserver.com:5060;transport=tcp> This is the full 200 response from FS: SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.0.1:52631;rport=42810;branch= z9hG4bKPjwPkMUAMcou1XPnDrrWJH9GDwdlaPKXCF;alias;received=2.3.4.5 From: "1020" <sip:1020@fs-dev1.sipserver.com>;tag=h9ydSS- jO2dQ7xT8yxpVcLu-appl6Xtl To: <sip:123123123@fs-dev1.sipserver.com>;tag=vXtN8yZgtHy0B Call-ID: Y3tWlOhNz3f9bt-uHO-LjFjh2RQwQh9l CSeq: 13167 INVITE Contact: <sip:123123123@1.2.3.4:5060;transport=tcp> //HOW TO GET THIS TO USE FQDN User-Agent: FreeSWITCH-mod_sofia/1.6.16~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 248 Remote-Party-ID: “123123123" <sip:123123123@fs-dev1.sipserver.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1522636701 1522636702 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 18966 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=rtcp:18967 IN IP4 1.2.3.4 Any ideas please? Thanks!

2 1

Question about limit max_calls
by Vlad Toma 09 Apr '18

09 Apr '18

Hello, I want to restrict the number of incoming calls to my internal extensions to 1 per extension, because I don’t want to receive another call to an extension that is already in a call. I tried setting the max_calls variable to 1 and adding this to the dialplan <extension name="limit_exceeded"> <condition field="destination_number" expression="^limit_exceeded$"> <action application="playback" data="/sounds/overthelimit.wav"/> <action application="hangup"/> </condition> </extension> <extension name="limit" continue="true"> <condition>  <action application="set" data="max_calls=1" inline="true"/> <action application="limit" data="db $${domain} ${sip_auth_username} ${max_calls}"/> </condition> </extension> Also I tried adding this to my dialplan where the condition matches my internal extensions <action application="limit_execute" data="hash outbound carrier1 1 bridge {ignore_early_media=true}user/${dialed_extension}@${domain_name}" /> It works if I dial normally but if I do a conference conf dial {ignore_early_media=true,conference_member_flags=moderator}user/XXX@${domain_name} it always calls even if that user is already in a call. How can I fix this? Thank you. Kind regards, Vlad Toma

2 3

Older OpenVox G400E card
by Peter Nerád 09 Apr '18

09 Apr '18

Hi it is possible to use older openvox G400E card with the latest freeswitch ? Where to start to get this card to live ? :-D Thanks for any help

2 1

Controlling an inbound call remotely over ESL - dialplan question
by Andrew Keil 09 Apr '18

09 Apr '18

To FreeSWITCH users, I would like to control an inbound call to FreeSWITCH via "inbound ESL to FreeSWITCH" and I need to know how I can write a simple diaplan which basically does nothing (just sits there and does not even answer the caller and does not hangup on the caller). Then I should be able to control the entire call remotely over ESL (even call answer). Maybe I do not even need a dialplan? Please note that I am wishing to use "inbound ESL to FreeSWITCH" to control the call, not outbound ESL, and I fully understand how to send API commands over ESL. Andrew

2 2

how to access the XML_REQUEST in mod_python
by Do Nguyen Ha 09 Apr '18

09 Apr '18

hi list i want to use mod_python to dynamic load user directory and read the wiki but no documents for access XML_REQUEST how to get the XML_REQUEST in python script thank you

2 4

Adding a Parked Call into a conference
by Bob McCarthy 07 Apr '18

07 Apr '18

Anyone have a best practice for taking a channel out of a parking lot and adding it to a conference? I have successfully transferred into the parking lot but the results are less than desired. The BLF remains active until the conference ends and pressing it give you hold music. Looking for an API / lua solution. Thanks in advance, Bob McCarthy

2 2