July 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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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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SIP Via Header on 100 Trying
by Grant Bagdasarian 15 Jul '20

15 Jul '20

Hi, I came across a known issue, and am wondering about the progress of https://github.com/signalwire/freeswitch/issues/319. One of our carriers is requiring the 2nd Via header to be sent back when Freeswitch triggers the 100 Trying. I tried to disable auto-invite-100 and use the acknowledge_call application in the dialplan, but without success. Is there a known workaround? Regards, Grant

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Optional Approach for SRTP (rfc 8643)
by Eugene Christensen 14 Jul '20

14 Jul '20

Hello, I am attempting to get optional SRTP to work based on rfc 8643. I have found that I can configure FreeSWITCH to allow for crypto in RTP/AVP, but I don't seem to get it to work fully. I've used the configuration elements rtp_allow_crypto_in_avp and NDLB-allow-crypto-in-avp. Both give me the same results but neither fully works. I found in the archives another person who experienced this same issue. It was in October of 2018 and is titled 'Accepting an "optional" SRTP offer (crypto in RTP/AVP) and establishing SRTP'. In that case, he was using version 1.6.20. From the responses, it seemed like in 1.8.2 it was possibly supposed to be working. I'm using 1.8.5 and I have tried it in 1.10.3 as well but they both seem to behave the same. A recap of what is in the SDP offer and the SDP answer are as follows: The endpoint is signaling optional SRTP according to rfc 8643 by sending a crypto token or a fingerprint in the SDP for a media descriptor that is signaled as RTP/AVP. m=audio 58702 RTP/AVP 0 8 101 c=IN IP4 65.37.249.200 b=AS:64 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:30 a=crypto:1 AES_256_CM_HMAC_SHA1_80 inline:8u1pUNOnFJUWKAa25GEtyFnHebV6AHrdpDgZ47nYBU6AtxlWiECv+RZPOSjW5g== a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:q3iDyAquDcVye5MVFF4NR2Krli3PdmSwxMKGFK8D a=setup:active a=fingerprint:sha-1 5F:29:44:20:8A:CB:81:1C:DC:9B:B0:C6:4D:EB:01:C1:6B:B8:CE:8E I can see in the console log that the initially prepared SDP answer contains an m-line with RTP/SAVP and it also contains the selected crypto attribute for the response. m=audio 25190 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=sendrecv a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:JfypK+0Vu1y6WXx0B/hhARVcvzYmO0Tt3xRJGx3o Yet before it is sent, it is changed to have the media channels disabled. m=audio 0 RTP/AVP 19 What else do I need to do to get this to work with optional encryption coming in the SDP-Offer? Thanks. Eugene Christensen CONFIDENTIALITY NOTICE. This e-mail transmission, and any documents, files or previous e-mail messages attached to it, may contain confidential and proprietary information. If you are not the intended recipient, or a person responsible for delivering it to the intended recipient, you are hereby notified that any disclosure, copying, distribution or use of any of the information contained in or attached to this message is STRICTLY PROHIBITED. If you have received this transmission in error, please immediately notify me by reply e-mail at echristensen(a)sorenson.com<mailto:echristensen@sorenson.com> or by telephone at +1 (801) 287-9419, and destroy the original transmission and its attachments without reading them or saving them to disk.

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Should I use loopback for a hunt group, or is there a better way?
by Martin Paterson 13 Jul '20

13 Jul '20

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 want. 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 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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fs_cli timeout on centos7
by Vlad Toma 13 Jul '20

13 Jul '20

Hello, I have 3 centos 7 servers with Freeswitch 1.10.2-release.4~64bit with core db on a 3 server galera cluster. My fs_cli randomly freezes when I issue commands such as Sofia status or show registrations, or any other commands. How can I debug this problem? Thank you. Regards, Vlad

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Sofia Profile - STUN Candidate without the "ext-sip-ip" setting is not working.
by Alexander Haugg 13 Jul '20

13 Jul '20

Hi, my problem is, I need the local SIP address and the external address for the candidates. The setting is vars.xml <X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/> Sofia profile: <param name="stun-enabled" value="true" /> <param name="stun-auto-disable" value="false" /> <param name="rtp-ip" value="1.2.3.4" /> <param name="sip-ip" value="1.2.3.4" /> <param name="ext-rtp-ip" value="$${external_rtp_ip}" /> <param name="ext-sip-ip" value="$${external_rtp_ip}" /> Call Cmd: originate {include_external_ip=true,media_webrtc=true}sofia/gateway/GW_SBC2_B2Bua/123 &park In this case the ICE candidates in de SDP are OK but the Contact header have the external sip address. If I do unset the "ext-sip-ip", then the Contact header is OK but I'm missing the external ICE candidates. What can I do that the STUN candidate will set to the SDP without the "ext-sip-ip" config? Thanks a lot Alex

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Call media stats in Freeswitch log
by Shahnawaj Khan 13 Jul '20

13 Jul '20

Hello All, Please let me know if it is possible to log media stats in freeswitch.log after call complete such as -<call-stats> -<audio> -<inbound> <raw_bytes>60716</raw_bytes> <media_bytes>60716</media_bytes> <packet_count>353</packet_count> <media_packet_count>353</media_packet_count> <skip_packet_count>234</skip_packet_count> <jitter_packet_count>0</jitter_packet_count> <dtmf_packet_count>0</dtmf_packet_count> <cng_packet_count>0</cng_packet_count> <flush_packet_count>0</flush_packet_count> <largest_jb_size>0</largest_jb_size> <jitter_min_variance>5.73</jitter_min_variance> <jitter_max_variance>28.03</jitter_max_variance> <jitter_loss_rate>0.00</jitter_loss_rate> <jitter_burst_rate>0.00</jitter_burst_rate> <mean_interval>20.00</mean_interval> <flaw_total>0</flaw_total> <quality_percentage>100.00</quality_percentage> <mos>4.50</mos> </inbound> -<outbound> <raw_bytes>97352</raw_bytes> <media_bytes>97352</media_bytes> <packet_count>566</packet_count> <media_packet_count>566</media_packet_count> <skip_packet_count>0</skip_packet_count> <dtmf_packet_count>0</dtmf_packet_count> <cng_packet_count>0</cng_packet_count> <rtcp_packet_count>0</rtcp_packet_count> <rtcp_octet_count>0</rtcp_octet_count> </outbound> </audio> </call-stats> or any other variable that is available in xml CDR Any insight is welcomed! Thanks & Regards, Shahnawaj Khan

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