- FreeSWITCH-users - FreeSWITCH Mailing Lists

Voicemail to email problem
by Vasyl Olenchenko 16 Jan '14

16 Jan '14

Hi. I have a problem with voicemail sent by e-mail. I do have followinf parameters specified: <param name="vm-email-all-messages" value="true"/> <param name="vm-mailto" value="xxx(a)xxx.yyy"/> <param name="vm-notify-mailto" value="xxx(a)xxx.yyy"/> <param name="vm-attach-file" value="true"/> <param name="vm-message-ext" value="wav"/> as well as: <param name="mailer-app" value="sendmail"/> <param name="mailer-app-args" value="-f xxx(a)xxx.yyy"/> in switch.conf.xml Also I can successfully send mail from command line (Im using ssmtp), but voicemail is not sent. I don't even have: [DEBUG] switch_utils.c:763 Emailed data to [xxx(a)xxx.yyy] [DEBUG] mod_voicemail.c:2881 Sending notify message to xxx(a)xxx.yyy In my logs - so it looks like Freeswitch is not even trying to send email. I have this in my logs: ... [DEBUG] mod_voicemail.c:1942 Update MWI: Voice Message 14/0 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] [DEBUG] switch_ivr_play_say.c:251 Handle play-file:[voicemail/vm-goodbye.wav] (en:en) [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@8000hz 1 channels 20ms [DEBUG] switch_ivr_play_say.c:1717 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-goodbye.wav [NOTICE] switch_core_state_machine.c:262 loopback/voicemail-b has executed the last dialplan instruction, hanging up. [NOTICE] switch_core_state_machine.c:264 Hangup loopback/voicemail-b [CS_EXECUTE] [NORMAL_CLEARING] Any suggestion is highly appreciated. Thanks in advance.?

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[gateways] remove/change "gw=********" in sip contact header (external gateway register)
by Douglas Adami 16 Jan '14

16 Jan '14

Hello guys, We can add new parameters by setting "contact-params", but how do I remove the default parameter "gw =" in sip contact header during the external gateway register? I do not want that information becomes available to the external server. what is the correct way to do it? Regards, Doug

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CHANNEL_HANGUP_COMPLETE cannot received
by 刘云龙 16 Jan '14

16 Jan '14

Hi! I'm using mod_erlang_event outbound, the env is FreeSWITCH Version 1.2.14 and Erlang R16B02, My erlang applicaton behaviour normal before the call was hangup, but it cannot receive every CHANNEL_HANGUP_COMPLETE, any one can tell me why?

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How to check if a channel was actually bridged after a bridge command
by Ali Pey 16 Jan '14

16 Jan '14

Hello, Is there a way to determine if the call was actually bridged after a bridge command? I want to send the call to voicemail if the bridge was rejected by the b-leg through call screening but take some other action if the call was actually bridged and then the b-leg hung up. Thanks, Ali Pey

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Cannot build v1.2.stable branch on OS X Mavericks
by mbo 15 Jan '14

15 Jan '14

Hello, I’ not able to build latest v1.2.stable branch on OSX Mavericks. All ports are updated. See output here http://pastebin.freeswitch.org/21838. Does anyone have the same problem? Thanks Markus

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Unable to make outbound calls
by Ian McMaster 15 Jan '14

15 Jan '14

Apologies for a simple newbie question. Following examples in the first two FreeSWITCH books, have been able to set up a SIP gateway, register, and receive calls. However I can't make calls. The dialplan is being entered, but 'hangs' at switch_channel.c:1055 New Channel sofia/external/16475551212 a) The provider name is "babytel". b) The dial plan bridge is to data="sofia/gateway/babytel/$1" (as per the books) c) The provider xml file is in conf/sip_profiles/external (as per the books) d) sofia status Name: external::babytel Type: gateway State: REGED Why is the bridge channel called "sofia/external/...."? And not "sofia/babytel/..."? Why does sofia status show "external::<provider>" (the books just show the provider name)? Can anyone explain why the bridge to the gateway does not occur? Thank you, Ian.

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Multiple extensions sharing presence
by Stephen Barry 15 Jan '14

15 Jan '14

Hello I am hoping for some advice on whether something is possible: I have created 3 directory extensions: 3862 (Desk Phone), 38621 (PC Softclient), 38622 (Mobile SoftClient), these all need to ring (at different times) when someone dials 3862, for this I have created a ring group and that is working well. Now I would like them all to share presence as all 3 extensions are effectively the same person. I have this working for inbound calls to 3862 answered on either 38621 or 38622 as I have overridden the presence_id in the dialstring for those extensions. Is there a way somehow to get a call made on extension 38621 to update 3862's presence to 'on the phone' as 3862 is being monitored by several phones using BLFs. I have had a good look around the docs, the only thing I can see is the presence map which may be what I am after, but I cannot find any explanation of what it does and how it works. Perhaps there is no way to do this, or a better way to do this. Any advice would most gratefully received. Thanks Steve

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News, Notes, and ClueCon Weekly! Today 1PM EST
by Ken Rice 15 Jan '14

15 Jan '14

Hey Guys Join us today for ClueCon Weekly at sip:888@conference.freeswitch.org 1PM EST. Today we¹ll have Richard ³Moose² Genthner with Symplicity presenting Skydas a FreeSwitch PHP API and Angular JS Mod_Callcenter Panel. In other news, for you guys running Debian, we¹ve added 2 new debian repo¹s for the 1.4beta branch and the Master branch from GIT. See https://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_pack ages on how to enable these and start using the packages! (we¹ll some open this up from Centos users with new yum repos as well. Watch for that in the near future!) Have a great week FreeSWITCHers! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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JOB_UUID
by Ryan Gholam 15 Jan '14

15 Jan '14

Hey all , i have a question that kept me searching for a while how could i filter events concerning JOB-UUID if i have 100 JOBS

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FS verision 1.5.7b - sound is somewhat distorted
by Fredrik Åhlén 15 Jan '14

15 Jan '14

Hi everyone, this is my first exeperience with FreeSwitch. I just downloaded FreeSwitch and installed it on Windows 7 using the MSI-installer. I think it’s verision 1.5.7b, at least according to the output from fs_cli.exe. The installation is pure local on my own home LAN (100 mbit), i.e no VOIP gateways etc. I am playing around with the installation and the FSClient 1.2.3.5 as softphone calling the echo test, calling between extension 1000 – 1001, voice mail 4000 etc. Everything seems fine but the sound is somewhat distorted. The sound is a little/somewhat scratchy. I’ve tried to change codec in FreeSwitch and/or the FSClient but without luck. Since I am completely new to FreeSwitch I thought that I would ask the community about it. What might cause this sound distorsion? I have recorded the sound but I was not sure it I should attatch it or not. Any help is appreciated. Output from fs_cli.exe: “ freeswitch@internal> version FreeSWITCH Version 1.5.7b+git~20131111T224713Z~5456daa010~32bit (git 5456daa 201 3-11-11 22:47:13Z 32bit) “ >From vars.xml: “ <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> “ With Best Regards, Fred

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