September 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

grandstream GXW4104 (GXW410x) errors
by Karl Schmidt 25 Sep '13

25 Sep '13

I have the GXW4104 up and working - but I'm getting a error level log listed as: Sep 19 15:16:03 gxw4104 GS_LOG: [00:0B:82:48:E3:94][712][9660000322A][1.4.1.4] 1376 cm_rtp.c discard it, RTP Ver no match ! Sess: 0 Any clue as to what it might mean? Didn't see anything in the freeswitch logs at error level. At boot time the GXW4104 spits out a different error Sep 19 15:18:40 gxw4104 GS_LOG: [00:0B:82:48:E3:94][707][9660000322A][1.4.1.4] TIMER: Invalid timer period: 0, TimerID: 6 sip_transport.c:539 Sep 19 15:18:40 gxw4104 GS_LOG: [00:0B:82:48:E3:94][707][9660000322A][1.4.1.4] TIMER: Invalid timer period: 0, TimerID: 2 sip_transport.c:539 I'm actually writing up a how-to-config for FS on these. The SIP settings can be seen at http://xtronics.com/uploads/gxw4104_accounts.sip_settings.png -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Full government control of all activities of the individual is virtually the goal of both national parties. --Ludwig von Mises --------------------------------------------------------------------------------

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What is conference floor
by Rafal Gwizdala 25 Sep '13

25 Sep '13

... and how can it be used? R

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Gsmopen: K3520 automatic device discovery
by Johannes Stads 25 Sep '13

25 Sep '13

Hello. I've built FS on a CentOS 6.4 virtual machine (MacOS 10.8 Parallels) using minimal config for testing with Huawei K3520 dongle. It was initially unlocked (Vodafone DE) and voice enabled by default. Using manual USB port config all worked fine including calls and sms, voice quality was good. It uses USB2 port for modem, and USB1 for audio. I tried to attach 2 dongles, made used rules and that worked fine too, however this scenario does not guarantee the same order in which dongles are seen after each reboot. Automatic device discovery does not seem to work with that dongle. I specified IMSI for each dongle, or IMEA, or both. Every time mod_gsmopen tries to use /dev/ttyUSB3 which is not a proper choice for those dongles. I can see K3520 in the list of supported dongles. Maybe something I do wrong. Any idea? Jo

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Freeswitch Port
by Andre 25 Sep '13

25 Sep '13

What is the variable to find the SIP port in a dial plan? I'm looking for the 5060 port (assuming I didn't change it in the profile).

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"Upgrade" from 1.3b to 1.2stable caused higher cpu usage
by Freddy Andersen 25 Sep '13

25 Sep '13

Hi, We have 5 Freeswitch servers running with about 500 concurrent calls total so about 100 each at peak. These servers was installed with version FreeSWITCH Version 1.3.13b+git~20130215T194756Z~9733740cab (git 9733740 2013-02-15 19:47:56Z) and they have been running fine with 250% cpu (out of 2400%), load average hits 3 at peak. We wanted to get our Freeswitch onto a stable release and thought 1.2.12 was the most stable version so this is what we installed on one server. This servers cpu is at 800-900% and load average hits 10 when the server runs with 1.2.12, if we rollback cpu goes back to old 2-300%.. Call quality is still ok but I feel we have less room for more load and upgrading should normally improve performance not decrees performance. Just as an experiment I also tried to install latest 1.4 thinking this might have been something in the 1.2 branch but 1.4 acts the exact same. here is a quick vmstat snapshot: procs -----------memory---------- ---swap-- -----io---- -system-- ----cpu---- r b swpd free buff cache si so bi bo in cs us sy id wa 15 2 0 430144 321280 9988432 0 0 3 33 1 1 11 3 87 0 7 0 0 425160 321344 9991936 0 0 56 116 67726 113824 30 4 66 0 22 0 0 424380 321368 9993580 0 0 28 0 84236 148377 25 4 70 0 19 0 0 423656 321404 9994324 0 0 32 0 61714 103954 23 2 74 0 9 0 0 422740 321444 9996020 0 0 24 1524 52116 86355 23 2 74 0 7 0 0 421908 321456 9997160 0 0 20 5524 58977 98786 22 2 76 0 10 0 0 420812 321464 9998244 0 0 4 0 52964 83640 23 2 75 0 21 0 0 417152 321496 9999184 0 0 56 36 73430 124669 25 3 71 0 6 0 0 417496 321516 10000704 0 0 20 0 78778 134441 27 3 69 0 This is from sometime, same calls but from a freeswitch 1.3 server: procs -----------memory---------- ---swap-- -----io---- -system-- ----cpu---- r b swpd free buff cache si so bi bo in cs us sy id wa 3 0 244 2668752 269200 4589652 0 0 7 60 1 0 14 3 83 0 4 0 244 2666192 269228 4590676 0 0 64 12 44941 50134 24 4 72 0 3 0 244 2670200 269256 4592680 0 0 28 24 62350 86122 34 4 62 0 4 0 244 2668584 269280 4593848 0 0 20 0 43495 53052 27 2 70 0 6 0 244 2664636 269332 4595884 0 0 32 5576 53941 67240 26 4 70 0 8 0 244 2665316 269380 4596720 0 0 40 68 66434 91266 31 4 64 0 3 0 244 2666780 269404 4597792 0 0 24 0 41862 48055 23 3 74 0 -- - Freddy

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Re: [Freeswitch-users] inherit_codec failing with pre-media
by sangdrax8 25 Sep '13

25 Sep '13

Did I explain a bit to much detail for the users list? Does anyone use "inherit_codec=true" and have it working? On Tue, Sep 17, 2013 at 2:27 PM, sangdrax8 <sangdrax8(a)gmail.com> wrote: > I pulled the latest freeswitch from git this morning, and am still > having issues using inherit_codec=true. It seems the only way I can > get it to defer to the choice of the b-leg call is to remove the > ringback, and thus the need for early media. > > My setup has 2 phones with the following codecs configured: > > phone1000: g722, ulaw > phone1001: ulaw > > My goal is to have a call initiated by phone1000 (a-leg) to defer the > codec choice until phone1001 (b-leg) chooses one. This would allow > them to talk with no transcoding required by freeswitch. With the > following dial plan, this fails to work. > > <extension name="Local_Extension"> > <condition field="destination_number" expression="^(10[01][0-9])$"> > <action application="export" data="dialed_extension=$1"/> > <action application="set" data="ringback=${us-ring}"/> > <action application="set" data="transfer_ringback=$${hold_music}"/> > <action application="set" data="call_timeout=30"/> > <action application="set" data="hangup_after_bridge=true"/> > <action application="set" data="inherit_codec=true" /> > <action application="bridge" > data="user/${dialed_extension}@${domain_name}"/> > </condition> > </extension> > > It does log that it is going to defer: > > 2013-09-17 13:15:10.956535 [DEBUG] mod_sofia.c:4542 [zrtp_passthru] > Setting a-leg inherit_codec=true > 2013-09-17 13:15:10.956535 [DEBUG] mod_sofia.c:4545 [zrtp_passthru] > Setting b-leg absolute_codec_string='G722@8000h@20i@64000b,PCMU@8000h@20i@64000b' > > > then it tries to provide the ringback for the a-leg, and it negotiates > to g722 since freeswitch supports that and it is the first match. > Later when the b-leg is answered, it also negotiates, this time to > ulaw as that is all b supports. My call completes, but freeswitch has > to transcode between the two. > > In an attempt to see inherit_codec work, I then commented out the > ringback line from the above dialplan. With only this one change, > freeswitch now waits until the b-leg negotiation, and passes that back > to the a-leg. The call completes and both handsets are using ulaw and > no transcoding is required. The problem being that the calling party > never gets a ring while waiting. > > I can't imagine that the only way to use inherit_codec=true is with > all ringing turned off. Can someone tell me if there is an additional > option I am missing? All documentation I read seems to indicate that > just adding "inherit_codec=true" will fix things, but even the > default.xml provided as an example uses ringback and therefore fails.

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Help with MOH
by JP 25 Sep '13

25 Sep '13

I am trying to play MOH on a leg after bridge. Here are the steps 1. I "originate" a call( Call-A ). 2. I then "bridge" it to another call (Call-B). I have set "park_after_bridge=true". 3. After a while, I "transfer" the Call-B to a dialplan extension where I just "playback" moh as follows... <action application="playback" data="local_stream://moh" /> The problem is that I don't hear the moh on leg B even though I observe that the calls are un-bridged. It looks like both the calls remain in park state. Any help appreciated. Thanks JP

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Incomplete Hangup(Customer/Caller Still Active)
by Arpit Gadle 25 Sep '13

25 Sep '13

Dear All, When an agent receives,an incoming call he hangsup/disconnect/ignore without answering. The call is clear at agent's end and he is ready for another call but the call is still active at caller/customer end. * *I tried disconnecting the active call by "fsctl hupall normal_clearing" but failed, I also tried transfering call to to other extension in dialplan and hangup from there but no result. Any help is appreciated. Thanks & Regards, Arpit Gadle

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No TALK and NOTALK events are fired
by mathieu virard 24 Sep '13

24 Sep '13

Hi, I'm trying to get TALK and NOTALK events but it seems that theses events are never fired. That's what I've done so far: - add the line "<param name="vad" value="both"/>" under settings in my sip profile - add the lines "<action application="set" data="fire_talk_events=true"/> and <action application="set" data="fire_not_talk_events=true"/> " to my dialplan (I also tried with "export" instead of "set", and i tried to use the "rtp_enable_vad_out" variable without success) - I started a the fs_cli.exe console and typed: "/event talk" and "/event notalk" - I established a call which uses the modified sip profile and dialplan. I alternatively speak on the microphone and mute it, but i can't see any event on the console but the hearbeats (i also activate "/event heartbeat" to be sure that event are well displayed) Has someone an idea about what i did wrong / missunderstood? My intention is to detect inactivity on the call and to process this event in an external application connected to freeswitch. Thanks in advance for any help! Cheers, mathieu

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Issue with auth-calls false
by Trevor Benson 23 Sep '13

23 Sep '13

I am having trouble creating an unregistered external gateway in FS. I went through the wiki and found that auth-calls and ACL's looks like the proper way to achieve this. I set auth-calls to false in both internal & external, added an ACL into domains to allow the cidr of the gateway in question. However after the gateway sends the Invite, FS always sends a 407 Proxy Auth back to the gateway. I checked into the IRC channel and one or two nice folks reviewed the config and said they thought it should be working, but the call still fails on Proxy Auth 407. The SIP Trace, external, internal & gateway profile from the test system are available at http://pastebin.freeswitch.org/21443 All suggestions are welcome, Trevor

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