- FreeSWITCH-users - FreeSWITCH Mailing Lists

in 20 min!!! SIP Chronicles #7, Featuring Ken Rice
by Giovanni Maruzzelli 18 Jul '20

18 Jul '20

Hey, OpenSource folks ! this is just a friendly reminder. If you liked any of our first 6 episodes of SIP Chronicles, please consider putting some time aside this Saturday at 4:30 UTC to see Ken Rice talking about FreeSWITCH, OpenSIPS and other projects that he is working on. https://youtu.be/176gzKQOoSQ See you soon! -Max & Giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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SIP Chronicles #7, Featuring Ken Rice
by Maxim Sobolev 18 Jul '20

18 Jul '20

Hey, OpenSource folks this is just a friendly reminder. If you liked any of our first 6 episodes of SIP Chronicles, please consider putting some time aside this Saturday at 4:30 UTC to see Ken Rice talking about FreeSWITCH, OpenSIPS and other projects that he is working on. https://youtu.be/176gzKQOoSQ See you soon! -Max

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Is this an appropriate use of loopback or is there a better way?
by Martin Paterson 17 Jul '20

17 Jul '20

Hello all, I'm about to use loopback, but given the warnings against its use in teh docs I wondered if there was another way to achieve the behaviour I would like. I'm making a hunt group: several phones called at the same time - thats straightforward: create an extension in the dialplan which matches on the hunt group's number, put a bridge command inside, the dialstring contains the user agents to dial: {ignore_early_media=true}user/833@mydomain,user/835@mydomain But what if the hunt group wants to have another hunt group as a destination? I can do it with loopback within the bridge: {ignore_early_media=true}user/833@mydomain,user/835@mydomain:_:{ignore_early_media}loopback/202/mydomain But does anyone know of a better way? In other words can one do a combination of bridge and transfer in one go? Can I bridge/transfer to multiple destinations at once, some of which are user agents or external destinations (user/833 or sofia/gateway/my-gw/0123456789) and some of which are extensions in the dialplan? Many thanks, Martin. This message has been scanned by technologywithin Mail Control. Martin Paterson Development Team Phone: 0207 953 8840 Email: martin.paterson(a)technologywithin.com Chevron Business Park, Limekiln Lane, Southampton, Hampshire, SO45 2QL Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K Registered in England No: 5964349 | VAT Number: GB 902 5369 37

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Is this an appropriate use of loopback or is there a better way?
by Martin Paterson 17 Jul '20

17 Jul '20

Hello all, I'm about to use loopback, but given the warnings against its use in teh docs I wondered if there was another way to achieve the behaviour I would like. I'm making a hunt group: several phones called at the same time - thats straightforward: create an extension in the dialplan which matches on the hunt group's number, put a bridge command inside, the dialstring contains the user agents to dial: {ignore_early_media=true}user/833@mydomain,user/835@mydomain But what if the hunt group wants to have another hunt group as a destination? I can do it with loopback within the bridge: {ignore_early_media=true}user/833@mydomain,user/835@mydomain:_:{ignore_early_media}loopback/202/mydomain But does anyone know of a better way? In other words can one do a combination of bridge and transfer in one go? Can I bridge/transfer to multiple destinations at once, some of which are user agents or external destinations (user/833 or sofia/gateway/my-gw/0123456789) and some of which are extensions in the dialplan? Many thanks, Martin.

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Should I use loopback to make a hunt group, or is there a better way?
by Martin Paterson 17 Jul '20

17 Jul '20

Hello all, I'm about to use loopback, but given the warnings against its use in the docs I wondered if there was another way to achieve the behaviour I would like. I'm making a hunt group: several phones called at the same time - that's straightforward: create an extension in the dialplan which matches on the hunt group's number, put inside it a bridge command, the dialstring contains the user agents to dial: "{ignore_early_media=true}user/833@mydomain,user/835@mydomain" But what if the hunt group wants to have another hunt group as a destination? I can do it with loopback within the bridge: "{ignore_early_media=true}user/833@mydomain,user/835@mydomain:_: {ignore_early_media=true}loopback/202/mydomain" But does anyone know of a better way? Another way of describing this would be: can one do a combination of bridge and transfer in one go? Can I bridge/transfer to multiple destinations at once, some of which are user agents or external destinations (user/833, or sofia/gateway/my-gw/0123456789) and some of which are extensions in the dialplan? Many thanks, Martin.

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Audio call get converted to video call after call is bypassed using uuid_media_3p API
by sagar malam 17 Jul '20

17 Jul '20

Hello All , Polycom VVX 500 series phones to make audio call. After call is answered i am executing uuid_media_3p API to bypass freeswitch , at this point audio call get converted to video call. I have observed that it is happening because polycom offers video codec in response to bypass REINVITE from free switch. So it is not FS issue but i want to know if there is anyway to remove video codecs before passing them to other leg. I have tried setting absolute codec string, codec string variables before bypass but they did not help. -- Thanks, Sagar

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Should I use loopback for a hunt group, or is there a better way?
by Martin Paterson 16 Jul '20

16 Jul '20

Hello all, I'm about to use loopback, but given the warnings against its use in the docs I wondered if there was another way to achieve the behaviour I would like. I'm making a hunt group: several phones called at the same time - that's straightforward: create an extension in the dialplan which matches on the hunt group's number, put inside it a bridge command, the dialstring contains the user agents to dial: "{ignore_early_media=true}user/833@mydomain,user/835@mydomain" But what if the hunt group wants to have another hunt group as a destination? I can do it with loopback within the bridge: "{ignore_early_media=true}user/833@mydomain,user/835@mydomain:_: {ignore_early_media=true}loopback/202/mydomain" But does anyone know of a better way? Another way of describing this would be: can one do a combination of bridge and transfer in one go? Can I bridge/transfer to multiple destinations at once, some of which are user agents or external destinations (user/833, or sofia/gateway/my-gw/0123456789) and some of which are extensions in the dialplan? Many thanks, Martin.

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Should I use loopback for a hunt group, or is there a better way?
by Martin Paterson 16 Jul '20

16 Jul '20

Hello all, I'm about to use loopback, but given the warnings against its use in the docs I wondered if there was another way to achieve the behaviour I would like. I'm making a hunt group: several phones called at the same time - that's straightforward: create an extension in the dialplan which matches on the hunt group's number, put inside it a bridge command, the dialstring contains the user agents to dial: "{ignore_early_media=true}user/833@mydomain,user/835@mydomain" But what if the hunt group wants to have another hunt group as a destination? I can do it with loopback within the bridge: "{ignore_early_media=true}user/833@mydomain,user/835@mydomain:_: {ignore_early_media=true}loopback/202/mydomain" But does anyone know of a better way? Another way of describing this would be: can one do a combination of bridge and transfer in one go? Can I bridge/transfer to multiple destinations at once, some of which are user agents or external destinations (user/833, or sofia/gateway/my-gw/0123456789) and some of which are extensions in the dialplan? Many thanks, Martin. Martin Paterson Development Team Phone: 0207 953 8840 Email: martin.paterson(a)technologywithin.com Chevron Business Park, Limekiln Lane, Southampton, Hampshire, SO45 2QL Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K Registered in England No: 5964349 | VAT Number: GB 902 5369 37

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Use of limit app
by Igor Potjevlesh 16 Jul '20

16 Jul '20

Hello there! I'm a bit confuse on how to properly use limit app to limit the call attempts per second. I have a customer with 3 IP addresses trusted and declared into an external SIP profile. All these 3 IPs are sending their calls to 1 IP on the Freeswitch. Would it be possible to limit the flow on the destination IP (which is the sip-ip or ext-sip-ip in their profile)? The first thing I found and which seems to work, is doing that based on their IP, but it's divided by 3 (because they have 3 IP). <action application="limit" data="hash ${sip_received_ip} ${sip_received_ip} 10/1 !NORMAL_CIRCUIT_CONGESTION"/> I'm not clear with what are possible realm and resources. Thanks for your inputs! Regards, Igor. -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

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Registering with external SIP provider
by LuKaRo 15 Jul '20

15 Jul '20

Hi everyone, I'm unable to register freeswitch with an external SIP provider. I gave the pages on sofia on the wiki already a good read, but I can't get it to work. One special thing is that my machine has multiple network interfaces and needs to use a specific one to connect to the external SIP provider. Therefore, I copied "external.xml" in "/opt/freeswitch/conf/sip_profiles" to "my-sip.xml" and changed the values for "rtp-ip", "sip-ip", "ext-rtp-ip" and "ext-sip-ip" to the IP address of the interface freeswitch should use, and added the "my-sip" directory to the "<X-PRE-PROCESS cmd="include" header. Then I created "/opt/freeswitch/conf/sip_profiles/my-sip/my-sip.xml" with the following contents: <include> <gateway name="my-telephony"> <param name="proxy" value="<ip address of SIP provider>"/> <param name="username" value="49xxxxxxxxxx"/> <param name="extension" value="49xxxxxxxxxx"/> <param name="password" value="passwordisnotneeded"/> <param name="register" value="true"/> <param name="register-transport" value="tcp"/> <param name="context" value="public"/> </gateway> </include> "sofia status" on the fs_cli lists the profile as "REGED": my-sip profile sip:mod_sofia@<local ip address>:5060 RUNNING (0) my-sip::my-sip gateway sip:49xxxxxxxxxx@<ip address of SIP provider> REGED ss -an|grep <ip address of SIP provider> also lists two established TCP-connections: tcp ESTAB 0 0 <local ip address>:46878 <ip address of SIP provider>:5060 tcp ESTAB 0 0 <local ip address>:5060 <ip address of SIP provider>:25976 However, "sofia status profile my-sip reg" shows 0 registrations: Registrations: ================================================================================================= Total items returned: 0 ================================================================================================= And when issuing a call from outside to the extension registered at the SIP provider, I just get a busy signal on the phone. Logs of the external SIP provider report "recipient unavailable". Any ideas what could be wrong or how to further diagnose this issue? How can I check if the call reaches freeswitch? Thanks a lot in advance, LuKaRo

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