November 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Server death bug on call origination
by Борисов, Дмитрий 24 Nov '14

24 Nov '14

Hi, All! When I do next: Dialplan: sofia/internal/104(a)company1.XXX.com Action answer() Dialplan: sofia/internal/104(a)company1.XXX.com Action bridge([presence_id= 101@company1.XXX.com]sofia/internal/sip:101@XXX.XXX.XXX.98 :1966;rinstance=91c49290308619e2:_:[presence_id=102@company1.XXX.com ]sofia/internal/sip:102@XXX.XXX.XXX.98:15582;rinstance=d04d400035af5525) my server is died with 100% CPU load and I will able recover it only by hardware reset. trying this on v1.4.9 and master current bordmi@switch:~ % uname -a FreeBSD switch.XXX.com 10.0-RELEASE-p10 FreeBSD 10.0-RELEASE-p10 #0: Mon Oct 20 12:42:25 UTC 2014 root@amd64-builder.daemonology.net:/usr/obj/usr/src/sys/GENERIC amd64 freeswitch@internal> version FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit) Virtual hardware by VMWare vSphere -- Regards, Dmitriy Borisov

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Re: [Freeswitch-users] Provisioning fifo.conf via xml_curl
by Dan Christian Bogos 24 Nov '14

24 Nov '14

Guys, Many thanks for your answers. @Vik: Yes, I am having the xml_curl correctly configured (I provision almost all, dialplan, directory, config, phrases in this way, and no issues elsewhere). @Steven: I thought I do. If I use exactly the same configuration received from xml_curl in fifo.conf, I have no issues. Bellow you can find what I am returning to FS, maybe you can spot some error or namespace issue? @Michael: I am loading mod_xml_curl very early at start (right after mod_console and mod_syslog) and mod_fifo way later. I thought that if I ask for a sample maybe I can spot the mistake (easy one I have imagined it) myself. Bellow you can find the configuration retrieval when I use console command "reload mod_fifo" Thanks again! DanB """ T 2014/11/24 16:22:18.818134 192.168.50.136:53681 -> 192.168.50.136:2880 [AP] POST /config/ HTTP/1.1. User-Agent: freeswitch-xml/1.0. Host: 192.168.50.136:2880. Accept: */*. Content-Length: 107. Content-Type: application/x-www-form-urlencoded. . hostname=nl-asd-dev-sbc01&section=configuration&tag_name=configuration&key_name=name&key_value=spandsp.conf ## T 2014/11/24 16:22:18.818764 192.168.50.136:2880 -> 192.168.50.136:53681 [AP] HTTP/1.1 200 OK. Date: Mon, 24 Nov 2014 15:22:18 GMT. Content-Length: 105. Content-Type: text/plain; charset=utf-8. . <document type="freeswitch/xml"><section name="result"><result status="not found" /></section></document> ###### T 2014/11/24 16:22:18.819212 192.168.50.136:53682 -> 192.168.50.136:2880 [AP] POST /phrases/ HTTP/1.1. User-Agent: freeswitch-xml/1.0. Host: 192.168.50.136:2880. Accept: */*. Content-Length: 107. Content-Type: application/x-www-form-urlencoded. . hostname=nl-asd-dev-sbc01&section=configuration&tag_name=configuration&key_name=name&key_value=spandsp.conf ## T 2014/11/24 16:22:18.819931 192.168.50.136:2880 -> 192.168.50.136:53682 [AP] HTTP/1.1 200 OK. Date: Mon, 24 Nov 2014 15:22:18 GMT. Content-Length: 105. Content-Type: text/plain; charset=utf-8. . <document type="freeswitch/xml"><section name="result"><result status="not found" /></section></document> ###### T 2014/11/24 16:22:18.820574 192.168.50.136:53683 -> 192.168.50.136:2880 [AP] POST /config/ HTTP/1.1. User-Agent: freeswitch-xml/1.0. Host: 192.168.50.136:2880. Accept: */*. Content-Length: 109. Content-Type: application/x-www-form-urlencoded. . hostname=nl-asd-dev-sbc01&section=configuration&tag_name=configuration&key_name=name&key_value=timezones.conf ## T 2014/11/24 16:22:18.820957 192.168.50.136:2880 -> 192.168.50.136:53683 [AP] HTTP/1.1 200 OK. Date: Mon, 24 Nov 2014 15:22:18 GMT. Content-Length: 105. Content-Type: text/plain; charset=utf-8. . <document type="freeswitch/xml"><section name="result"><result status="not found" /></section></document> ###### T 2014/11/24 16:22:18.821345 192.168.50.136:53684 -> 192.168.50.136:2880 [AP] POST /phrases/ HTTP/1.1. User-Agent: freeswitch-xml/1.0. Host: 192.168.50.136:2880. Accept: */*. Content-Length: 109. Content-Type: application/x-www-form-urlencoded. . hostname=nl-asd-dev-sbc01&section=configuration&tag_name=configuration&key_name=name&key_value=timezones.conf T 2014/11/24 16:22:19.517231 192.168.50.136:53685 -> 192.168.50.136:2880 [AP] POST /config/ HTTP/1.1. User-Agent: freeswitch-xml/1.0. Host: 192.168.50.136:2880. Accept: */*. Content-Length: 104. Content-Type: application/x-www-form-urlencoded. . hostname=nl-asd-dev-sbc01&section=configuration&tag_name=configuration&key_name=name&key_value=fifo.conf ## T 2014/11/24 16:22:19.520502 192.168.50.136:2880 -> 192.168.50.136:53685 [AP] HTTP/1.1 200 OK. Date: Mon, 24 Nov 2014 15:22:19 GMT. Content-Length: 880. Content-Type: text/plain; charset=utf-8. . <configuration name="fifo.conf" description="FIFO Configuration"><fifos><fifo name="QUEUE1(a)sip.test.domain.com" importance="0"><member timeout="30" simo="1" lag="0">{sip_invite_domain=sip.test.domain.com,sip_h_X-AuthType=INTERNAL,sip_h_X-DomainTag=sip.test.domain.com,sip_h_X-EpType=SIP,sip_h_X-EpTag=endpoint2,sip_h_X-Destination=a41_u4_0_EPR,sip_h_X-Balancer=1.2.3.4,call_timeout=30}sofia/ipbxas/eprou-a41_u4_0_EPR@sip.test.domain.com;fs_path=sip:127.0.0.1</member><member timeout="30" simo="1" lag="0">{sip_invite_domain=sip.test.domain.com,sip_h_X-AuthType=INTERNAL,sip_h_X-DomainTag=sip.test.domain.com,sip_h_X-EpType=SIP,sip_h_X-EpTag=endpoint2,sip_h_X-Destination=a41_u205_0_EPR,sip_h_X-Balancer=1.2.3.4,call_timeout=30}sofia/ipbxas/eprou-a41_u205_0_EPR@sip.test.domain.com;fs_path=sip:127.0.0.1</member></fifo></fifos></configuration> """

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Re: [Freeswitch-users] uuid_send_dtmf problem with rfc2833
by Adam Kuśmirek 24 Nov '14

24 Nov '14

Michael, it is not nat problem. Freeswitch is not sending dtmf events. I can't see anything on local interface. Italo, thanks for workaround, i will try it and send feedback. Regards Adam

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RTP Packet Loss Troubleshooting
by Linux Vince 24 Nov '14

24 Nov '14

I have installed freeswitch and testing with SiPP FreeSwitch was easily able to handle 1700 concurrent calls. However when it starts going near 2000 my RTP packets are being lost and audio is totally gone in some cases. How do i troubleshoot this ? And how to tune RTP performance for FreeSwitch. I checked that CPU is not fully utilized and is somewhere around 40% range and memory is only at 10%. Also network interface is not fully utilized as well as i was able to transfer files from server and interface was able to handle much more bandwidth than it was running. Please help.

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Fwd: How to provide ringback tone while transferring a parked call
by Szeto, Steven 24 Nov '14

24 Nov '14

1. Suppose you originate a call and park it: originate user/1001 &park 2. Now transfer the parked call: uuid_transfer d7605992-7e0d-4c24-9634-75fe081422ff bridge:user/1002 inline When you do this, the parked caller hears nothing until the destination answers. How to you pass enough parameters to the uuid_transfer command to play ringback tone to the parked call during the transfer? -- This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.

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Provisioning fifo.conf via xml_curl
by Dan Christian Bogos 24 Nov '14

24 Nov '14

Hey Guys, Was wondering if someone could help me with a working fifo.conf provisioned over xml_curl. I am trying to provision it but for some reason my configuration is not picked up, moreover if I am deleting autoload_configs/fifo.conf the mod_fifo will not even be loaded (although I am pushing it via xml_curl). Thanks in advance for any kind of tip! DanB

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uuid_send_dtmf problem with rfc2833
by Adam Kuśmirek 24 Nov '14

24 Nov '14

Hello, I have a problem with sending dtmf events (rfc2833) with uuid_send_dtmf api command. FreeSWITCH Version 1.5.15b+git~20141118T231404Z~df423b88d6~64bit (git df423b8 2014-11-18 23:14:04Z 64bit) First i originate the call and park it with the command: originate {origination_caller_id_name=test}sofia/gateway/test/ xxxxxxxx(a)yyy.zzz.com &park() The other side answers the call. I can see that rfc 2833 is negotiated in SDPs. SIP/2.0 200 Ok Via: SIP/2.0/UDP 46.174.232.238:5070;rport;branch=z9hG4bKFmQvapcNNp44m From: "..." <sip:...@...>;tag=35FN6HXZ3ry2F To: <sip:12122777254@sip.12voip.com>;tag=f0313ac544a16a82aee0b Contact: sip:...@...:5060 Call-ID: f7b0d42d-ed32-1232-06a5-c03fd5651f3e CSeq: 68010140 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 211 v=0 o=xxx 1416695524 1416695524 IN IP4 yyyy s=SIP Call c=IN IP4 yyyy t=0 0 m=audio 27792 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 Then I try to send DTMF freeswitch@internal> uuid_send_dtmf 6022b884-7297-11e4-8682-fd8246740d0e 12345 -ERR no reply 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ 12122777254(a)sip.12voip.com send dtmf digit=1 ms=250 samples=2000 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ 12122777254(a)sip.12voip.com send dtmf digit=2 ms=250 samples=2000 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ 12122777254(a)sip.12voip.com send dtmf digit=3 ms=250 samples=2000 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ 12122777254(a)sip.12voip.com send dtmf digit=4 ms=250 samples=2000 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ 12122777254(a)sip.12voip.com send dtmf digit=5 ms=250 samples=2000 Unfortunatelly it doesn't work and I can't see any rtpevents in wireshark. If I change dtmf-type in profile to info, i can see sip info packets. Please help

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Strange behaviour when calling 2 destinations in parallel
by kthofer 23 Nov '14

23 Nov '14

Hi There one incomming call shall originate 2 outgoing calls in parallel. one outgoing channel shall be terminated to a SIP/PSTN gateway the other one shall be terminated to a sip extension connected to a OPensips server. Both extensions (sip and PSTN) are ringing but when one of them answers the call, I get the loose_race from the not answered channel and then all calls are being released and everything collapse. any idea what this can be? I uploaded the whole call on to past bin please can someone have a look what the reason can be that the calls are releasing. past bin link http://pastebin.freeswitch.org/23613 -- With best regards Karl Theo Hofer M: +46 7030 22178 E: karl-theo_hofer(a)inteli-sim.com

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mod_sofia.c:2219 CODEC NEGOTIATION ERROR
by youssef elouam 23 Nov '14

23 Nov '14

Dear All ; many thanks in advance for your help. I have problem of CODEC NEGOTIATION in my freeswitch installed on Raspberry PI board. following is logs I have : 2014-11-23 16:52:16.832177 [ERR] mod_sofia.c:2219 CODEC NEGOTIATION ERROR. SDP:v=0o=- 3625750336 3625750336 IN IP4 192.168.1.5s=pjmediac=IN IP4 192.168.1.5t=0 0m=audio 4006 RTP/AVP 99 0 8 101c=IN IP4 192.168.1.5a=rtpmap:99 SILK/24000a=fmtp:99 useinbandfec=0a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtcp:4007 IN IP4 192.168.1.5 freeswitch@internal> versionFreeSWITCH Version 1.5.15b+git~20141024T202644Z~12b6940644~32bit (git 12b6940 2014-10-24 20:26:44Z 32bit)

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Changing receiver IP on INVITE
by Kurtis Heimerl 23 Nov '14

23 Nov '14

Hey FreeSWITCH Users, Interesting issue here. We're connecting to Nexmo, a major SIP provider. They do load balancing of their SIP servers; a set of four IPs behind sip.nexmo.com. I have a FS sip set up to register to their server. However, in INVITE, it seems to switch IPs. Basically it goes like this: US -> INVITE sip.nexmo.com (IP1) IP1 -> 407 AUTH US -> INVITE sip.nexmo.com (DIFFERENT IP2) nonce and auth stuff IP2 -> 904 FAIL Here's my config: <include> <gateway name="sip.nexmo.com"> <param name="username" value="XXXXX"/> <param name="password" value="YYYYY"/> <param name="from-user" value="XXXXX"/> <param name="from-domain" value="sip.nexmo.com"/> <param name="proxy" value="sip.nexmo.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="extension" value="5000"/> ... How do we stop this from happening? Setting from-domain and proxy to an IP behind their load balancer fixes it, but that's clearly suboptimal. Why is FS resolving the sip.nexmo.com DNS twice? Thanks!

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