September 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

DTMF
by Andrew 16 Sep '14

16 Sep '14

Hi, I have a few questions. I found if I make a call and the call connects that DTMF and avmd does not work unless I play an audio file. Here are a few settings I am using. session.Execute("start_dtmf_generate", "true"); session.Execute("set", "sleep_eat_digits=true"); session.Execute("set", "suppress-cng=false"); I also found that sometimes when I play an audio file and if I am not talking that even the audio file will not play. If I talk the audio continues to play. Not sure if this is a bug or a settings. I also see that log AVMD using fs_cli.exe get output to the screen only when audio file is playing or me talking. (odd). What I want to do is make a call and hold for DTMF and or beep (AVMD) without having to play audio.

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Call Released with a 200 OK
by Stephen Wilde 16 Sep '14

16 Sep '14

I have a strange issue: some calls are dropped with a 200 OK containing the "Reason" header. Is this valid? The message flow is: FS ---- SUPPLIER INVITE -> <- 100 Trying <- 180 Ringing

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Using mod_com_amd
by Julien Coniglio 16 Sep '14

16 Sep '14

Hi all, I'm looking for some documentation about the commercial AMD mod for FreeSWITCH (bought on freeswitch.org) I've currently installed it, detected licence and it's available on my FreeSWITCH instance, but I didn't find any documentation how to use it in production. On FreeSWITCH wiki, there is only manual about the commercial G729 mod. Thanks in advance, Julie -- Julien CONIGLIO - Telco ------------------------------------- Télémaque SAS Les Espaces de Sophia - Bât. E 80 route des Lucioles BP 40139 06903 Sophia Antipolis CEDEX - France

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Not Able to Connect to mysql
by Aqs Younas 16 Sep '14

16 Sep '14

Hi, All I am unable to connect to mysql for querying some data that I could use in my python script. Here is my python script. #!/usr/bin/python import os import re import json import MySQLdb from freeswitch import * db = MySQLdb.connect("localhost","root","hammer","gvt") def handler(session, args): session.answer() callerNumber=session.getVariable("caller_id_number") calleeNumber=session.getVariable("destination_number" consoleLog("info", callerNumber) consoleLog("info", calleeNumber) When i run this i got this error. 2014-09-16 04:55:18.803406 [ERR] mod_python.c:239 Error reloading module 2014-09-16 04:55:18.803406 [ERR] mod_python.c:164 Python Error by calling script "load": <class '_mysql_exceptions.InternalError'> Message: (-1, 'server not initialized') Exception: None Traceback (most recent call last) File: "/usr/lib/python2.7/load.py", line 9, in <module> File: "/usr/lib/python2.7/dist-packages/MySQLdb/__init__.py", line 81, in Connect File: "/usr/lib/python2.7/dist-packages/MySQLdb/connections.py", line 179, in __init__ Could someone please help me out how to connect freeswitch to mysql for querying data..:) Thanks in Advance

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Re: [Freeswitch-users] skinny issue
by Nick Zaitsev 16 Sep '14

16 Sep '14

Thank you for your reply. So, is there any way to set different contexts for sccp users? Thnak you for your time. Mon, 15 Sep 2014 17:21:10 -0400 от Adolphe Cher-Aime <acheraime(a)gmail.com>: > > I confirm that setting context in there makes no effect. > > > > On Mon, Sep 15, 2014 at 5:03 PM, Nathan Neulinger < nneul(a)mst.edu > wrote: > >Actually, scratch that, I just had the <variables> section in the wrong place in the user directory entry when I tested it. > > > ><?xml version="1.0"?> > ><include> > > <user id="SEP00115Cxxxxxx"> > > <params> > > <param name="a1-hash" value="InvalidHash"/> > > </params> > > <variables> > > <variable name="SKINNY_TESTING" value="TESTING_VALUE"/> > > </variables> > > <skinny> > > <params> > > <param name="ext-pickup" value="pickup-NathanTest"/> > > </params> > > <buttons> > >...snip... > > </buttons> > > </skinny> > > </user> > ></include> > > > > > >That will succesfully set that variable for the outgoing calls. I don't know that it will actually work for setting a > >context though - likely not. > > > >-- Nathan > > > >On 09/15/2014 03:54 PM, Nathan Neulinger wrote: > >> It does indeed appear that this is broken. I thought I had implemented it, but it doesn't appear to be getting processed > >> currently, and hadn't noticed cause I had worked around the limitation before I implemented it. > >> > >> Please file a Jira and I'll look into as time permits. > >> > >> -- Nathan > >> > >> On 09/15/2014 10:45 AM, Nick Zaitsev wrote: > >>> > >>> I mean, that the setting any variable in the user directory has no effect. > >>> > >>> -- > >>> Nick Zaitsev > >>> > >>> > >>> _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting(a)freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://confluence.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >>> http://www.cudatel.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users(a)lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > > > >-- > >------------------------------------------------------------ > >Nathan Neulinger nneul(a)mst.edu > >Missouri S&T Information Technology (573) 612-1412 > >System Administrator - Architect > > > >_________________________________________________________________________ > >Professional FreeSWITCH Consulting Services: > >consulting(a)freeswitch.org > >http://www.freeswitchsolutions.com > > > >Official FreeSWITCH Sites > >http://www.freeswitch.org > >http://confluence.freeswitch.org > >http://www.cluecon.com > > > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server > >http://www.cudatel.com > > > >FreeSWITCH-users mailing list > >FreeSWITCH-users(a)lists.freeswitch.org > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > >http://www.freeswitch.org > > >

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Grandstream GXP2130 - Do not hangup Properly
by Lloyd Aloysius 16 Sep '14

16 Sep '14

Hi All I am having issues with Grandstream GXP2130. When we make any calls from this phone, either internal or external, Grandstream GXP2130 not hangup properly and hangup after time out. I see the SIP logs, phone is sending the BYE. ------------------------------------------------------------------------ recv 585 bytes from udp/[A.B.C.D]:1028 at 23:14:08.086342: ------------------------------------------------------------------------ BYE sip:235@I.J.K.L:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.240.131:5060;branch=z9hG4bK855972464;rport From: "Nelson Eddy" <sip:nelson.eddy@mydomain.ca>;tag=147586373 To: <sip:235@mydomain.ca>;tag=t6SeUBtDa9BQK Call-ID: 1863291899-5060-7(a)BJC.BGI.CEA.BDB CSeq: 52 BYE Contact: <sip:nelson.eddy@192.168.240.131:5060> X-Grandstream-PBX: true Max-Forwards: 70 Supported: replaces, path, timer User-Agent: Grandstream GXP2130 1.0.4.10 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 ------------------------------------------------------------------------ Does anyone have the similar issue. How to solve this issue Thank you LLoyd

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mod_gsmopen audio delay
by Daniel Ivanov 16 Sep '14

16 Sep '14

I am using a Huawei K3715 modem and the audio is great except that it comes after like 4-5 seconds after call establishment. Calls on a HW E173 are going smooth with same config from the second call is established. Has anyone experienced something similiar and point me to directions where to look at?

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Question about DTMF duration
by Grant Bagdasarian 16 Sep '14

16 Sep '14

Hello, I'm having a small issue with DTMF, but it's not related to Freeswitch but the endpoint where the DTMF is coming from. I hope someone could clear this out for me. The first debug example of DTMF received by Freeswitch seems too short, and also the application which is running on FS doesn't recognize it. The second debug example is much larger in output and also its duration. The application on FS does recognize it. The first example is coming from a machine which receives the call from ISDN and bridges the call to a SIP network. The second example is pure SIP all the way. Does this mean the DTMF conversion between ISDN and VOIP is broken or misconfigured? Example 1 (Bad): 2014-09-16 12:01:59.766110 [DEBUG] switch_rtp.c:2798 Send start packet for [1] ts=2098368502 dur=160/160/800 seq=47884 lw=2098368502 2014-09-16 12:01:59.786012 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2098368502 dur=320/320/800 seq=47885 lw=161 2014-09-16 12:01:59.806003 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2098368502 dur=480/480/800 seq=47886 lw=321 2014-09-16 12:01:59.826107 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2098368502 dur=640/640/800 seq=47887 lw=161 2014-09-16 12:01:59.846100 [DEBUG] switch_rtp.c:2698 Send end packet for [1] ts=2098368502 dur=800/800/800 seq=47888 lw=1 2014-09-16 12:01:59.846100 [DEBUG] switch_rtp.c:2698 Send end packet for [1] ts=2098368502 dur=800/800/800 seq=47889 lw=1 2014-09-16 12:01:59.846100 [DEBUG] switch_rtp.c:2698 Send end packet for [1] ts=2098368502 dur=800/800/800 seq=47890 lw=1 2014-09-16 12:01:59.846100 [DEBUG] switch_rtp.c:2651 Queue digit delay of 40ms Example 2 (Good): 2014-09-16 12:06:56.566019 [DEBUG] switch_rtp.c:2798 Send start packet for [1] ts=2128118524 dur=160/160/2400 seq=23409 lw=2128118524 2014-09-16 12:06:56.566019 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=320/320/2400 seq=23410 lw=2128118684 2014-09-16 12:06:56.586046 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=480/480/2400 seq=23411 lw=2128118844 2014-09-16 12:06:56.586046 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=640/640/2400 seq=23412 lw=2128119004 2014-09-16 12:06:56.606023 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=800/800/2400 seq=23413 lw=161 2014-09-16 12:06:56.606023 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=960/960/2400 seq=23414 lw=321 2014-09-16 12:06:56.626017 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=1120/1120/2400 seq=23415 lw=481 2014-09-16 12:06:56.626017 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=1280/1280/2400 seq=23416 lw=641 2014-09-16 12:06:56.646020 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=1440/1440/2400 seq=23417 lw=161 2014-09-16 12:06:56.646020 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=1600/1600/2400 seq=23418 lw=321 2014-09-16 12:06:56.666019 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=1760/1760/2400 seq=23419 lw=161 2014-09-16 12:06:56.666019 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=1920/1920/2400 seq=23420 lw=321 2014-09-16 12:06:56.686032 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=2080/2080/2400 seq=23421 lw=161 2014-09-16 12:06:56.686032 [DEBUG] switch_rtp.c:2698 Send middle packet for [1] ts=2128118524 dur=2240/2240/2400 seq=23422 lw=321 2014-09-16 12:06:56.706023 [DEBUG] switch_rtp.c:2698 Send end packet for [1] ts=2128118524 dur=2400/2400/2400 seq=23423 lw=1 2014-09-16 12:06:56.706023 [DEBUG] switch_rtp.c:2698 Send end packet for [1] ts=2128118524 dur=2400/2400/2400 seq=23424 lw=1 2014-09-16 12:06:56.706023 [DEBUG] switch_rtp.c:2698 Send end packet for [1] ts=2128118524 dur=2400/2400/2400 seq=23425 lw=1 2014-09-16 12:06:56.706023 [DEBUG] switch_rtp.c:2651 Queue digit delay of 40ms

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Re: [Freeswitch-users] skinny issue
by Nick Zaitsev 15 Sep '14

15 Sep '14

I mean, that the setting any variable in the user directory has no effect. -- Nick Zaitsev

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SCCP issue
by Nick Zaitsev 15 Sep '14

15 Sep '14

I have one more issue. if i dial the wrong number,even the channel closes i have ring sound on my side. The fs sends to the Cisco phone 2014-09-15 18:39:58.113675 [DEBUG] mod_skinny.c:1000 [SEPE8BA70FB628D:0 @ 10.4.33.14:29107] Sending SetRinger with Ring Type (RingOff), Mode (RingForever), Line Instance (0), Call ID (23) how can i change the RingOff? Thank you for your time, Nick

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