- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Freeswitch Week in Review (Master Branch) November 9th-15th
by Ken Rice 21 Nov '14

21 Nov '14

New Post on freeswitch.org from kathleen check it out at http://ift.tt/11M59RI Freeswitch Week in Review (Master Branch) November 9th-15th Hello, again. This week in the FreeSWITCH master branch we had 33 commits. Some of the new features for this week include: support for logging full timestamps with dialplan (defaults to old behavior unless requested), the addition of external_video_source to media handle and expose switch_core_media_start_video_thread() to start the core video thread for non-rtp based media, allow mod_http_cache to S3 services other than Amazon, and new configuration parameter add-variables-to-offer (default=false) (When true, all channel variables are included in the offer to rayo client in mod_rayo). New features that were added: f175c71 FS-6805 add support for logging full timestamps with dialplan, defaults to old behavior unless requested. Jira: http://ift.tt/11E7kpC fada4b8 FS-6977 Do not create freeswitch.serial if zrtp not enabled Jira: http://ift.tt/11E7kpG dd629c1 Add external_video_source to media handle and expose switch_core_media_start_video_thread() to start the core video thread for non-rtp based media 0eefdca FS-6947 Opus RTP payload fmtp settings ( maxaveragebitrate / maxplaybackrate ) Jira: http://ift.tt/11M58xc dd61232 FS-6979 Allow mod_http_cache to S3 services other than Amazon Jira: http://ift.tt/11E7i0R 826d428 FS-6992 Global configuration or maxplaybackrate and maxaveragebitrate from opus.conf.xml Jira: http://ift.tt/11M58xe e1c0ef5 New configuration parameter, add-variables-to-offer (default=false) (When true, all channel variables are included in the offer to rayo client in mod_rayo). Improvements in cross platform build supports: 0f8b993 Fix mod_say_es_ar Makefile.am 07030c6 Fix compiler warning on unmatched return type The following bugs were squashed: 2f1b12f OPENZAP-232 Check for digits received on sangoma isdn stack to avoid delaying moving to the ring state if all digits are received at once in overlap dialing mode. 6b8d5b2 Fix release guard timer check in freetdm 34cf3b9 FS-6980 #resolve Fix for crash when using native recording on recordstop Jira: http://ift.tt/11E7i0U ab24bde FS-5533 fix issue with busy signal being sent back to all shared lines instead of just the calling device Jira: http://ift.tt/11M58xi a279bf3 FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia Jira: http://ift.tt/1uLG9Ug 0c68bb6 FS-6957 Fix for muting issues in mod_conference Jira: http://ift.tt/1uLG9Ue 33d37ce PLIV-13 Disable SSLv3 in libcurl for mod_httapi dbc5571 FS-6983 Wrap new curl TLS macro usage with ifdefs Jira: http://ift.tt/11E7i0W 0699ea8 Fixes for various issues 87a4670 FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia Jira: http://ift.tt/1uLG9Ug 82aa331 FS-6531: #resolve set to tag on auto answer notify Jira: http://ift.tt/11M58xk 75473a7 FS-6531: #resolve set to tag on uuid_phone_event notify Jira: http://ift.tt/11M58xk 07c5cc1 FS-7003 Fix infinite loop when output sent to server without SSML configured and repeat-times=0 Jira: http://ift.tt/11E7kG0 The complete list of commits can be found here:2014_11_9-2014_11_16

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HA recovery of custom channels variables
by Danny Gershman 21 Nov '14

21 Nov '14

I'm using some custom channel variables for storage and I noticed that when I do a recover they are not preserved because they are populated to the table 'recovery'. Is there something that I'm not doing to make this work? --Danny G

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Freeswitch Week in Review (Master Branch) November 2nd-8th
by Ken Rice 21 Nov '14

21 Nov '14

New Post on freeswitch.org from kathleen check it out at http://ift.tt/1Hwbe4X Freeswitch Week in Review (Master Branch) November 2nd-8th Hello, again. This week in the FreeSWITCH master branch we had 61 commits. Some of the many new features that were added this week are: the addition of console log level in mod_managed, support for per-module references directory in mod_managed , added pure CreateStateHandlerDelegate in ManagedSession for native api usage in mod_managed , addition of unset app for the chatplan (and while you can already unset by calling set with var=, same with set in dialplan, this is a convenience function similar to our unset in dialplan in mod_sms), and added a command to compile non-minified js file for testing in mod_verto. New features that were added: 7c0cf50 Set managedlist to return the value to the API stream and the log in mod_managed 4037e78 Support per-module references directory in mod_managed 10ebeba Add console log level in mod_managed 33cb950 Added pure CreateStateHandlerDelegate in ManagedSession for native api usage in mod_managed 889b678 Added GetPtr to Util class for internal pointers extraction (very useful when using native api) in mod_managed 8478874 FS-6831 Addition of unset app for the chatplan and while you can already unset by calling set with var=, same with set in dialplan, this is a convenience function similar to our unset in dialplan in mod_sms [Jira: http://ift.tt/1Hwa0Xw] bc767bb Adding rfc6598.auto and adding rfc6598 space to nat.auto acl. This is the NAT444 carrier grade nat space. 1190e59 FS-6965 Set the Accept-Language into a variable when received in mod_sofia [Jira: http://ift.tt/1uLG7M3] 94278b5 FS-5159 Allow enter and exit sounds to interrupt the MOH in a wait_mod conference 1944f9a FS-6968 Changes to mod_fifo.c to add outbound_per_cycle_min in mod_fifo [Jira: http://ift.tt/1Hwa0XA] 0f2816d Add command to compile non-minified js file for testing in mod_verto The following bugs were squashed: f87c335 Fix for issue with tracking agent availability 8f3c157 FS-6957 Fix for muting issues upon joining a conference in mod_conference [Jira: http://ift.tt/1uLG9Ue] 831832c FS-6890 Fix for FreeSWITCH forgets to send RE-INVITES [Jira: http://ift.tt/1uLG9Ug] 4eb5b38 Fix bug where re-invites needlessly re-init the codec and jb a497169 FS-6890 Additional work for FreeSWITCH forgetting to send RE-INVITES in mod_sofia [Jira: http://ift.tt/1uLG9Ug] 9c1e603 FS-6954 #comment Fix for side effect of another fix for this bug. Please test this patch. [Jira: http://ift.tt/1yzQC5V] f66f2ca FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia [Jira: http://ift.tt/1uLG9Ug] 415f82f FS-6954 Work toward fix for Freeswitch adding additional m=audio on t.38 call in mod_sofia [Jira: http://ift.tt/1yzQC5V] 5ce5199 FS-6969 Fix for unwanted profile shutdown in mod_verto [Jira: http://ift.tt/1Hwa0XE] In terms of stability these were the use cases that were fixed: f3d089a Fix a crash when trying to remove a shadow directory in mod_managed Improvements in cross platform build supports: 5dee5ce FS-6953 Fix for build issues with OS X [Jira: http://ift.tt/1uLG9Um] a17be38 Add reconf target 300b8d8 FS-6973 Fix for changed Opal svn url breaking build of mod_opal [Jira: http://ift.tt/1Hwa1dY] The complete list of commits can be found here:2014_11_2-2014_11_9

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