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Cannot call OUT
by T Fred Farmington 08 Apr '15

08 Apr '15

My FS Server is working to: * Allow 'internal' extensions to call each other * Allow 'external' calls to be received and routed as intended But I cannot place a call from 'inside' to an 'outside' number In the file conf/sip_profiles/external/velocity_gateway.xml I have my Gateway defined Within that file I have defined: -<gateway name="velocity_outbound"> username password Proxy IP address register=false (not needed) And in conf/sip_profiles/external.xml I have the command <X-PRE-PROCESS cmd="include" data="external/velocity_gateway.xml"/> In my conf/dialplan/default/outbound_via_velocity.xml I have defined the 'extensions' local_call.velocity long_distance_call.velocity Among the other 'extension' parameters is includes: <action application="bridge" data="sofia/gateway/velocity_outbound/$1"/> In my conf/dialplan/default.xml I have the command: <X-PRE-PROCESS cmd = "include" data = "default/outbound_via_velocity.xml"/> As a result, within FS_CLI, when I do a sofia status I see the following: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6::velocity_outbound gateway sip:FreeSWITCH@velocity_outbound NOREG external profile sip:mod_sofia@192.168.nnn.nn:5080 RUNNING (0) external::velocity_outbound gateway sip:FreeSWITCH@velocity_outbound NOREG external::velocity_outbound gateway sip:ClancySystems@208.nn.nnn.nn NOREG 192.168.nnn.nn alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@192.168.nnn.nn:5060 RUNNING (0) ================================================================================================= BUT when I attempt to make a call to an 'outside' number I find the following..... In my log I see where the outbound 'extension' is found (PASS) [local_call.velocity] destination_number(9nnnnnnnnnn) =~ /^9(\d{10})$/ break=on-false and it utilized the defined Gateway (velocity_outbound) Action bridge(sofia/gateway/velocity_outbound/nnnnnnnnnn) >From that point in the log, things continue, but do not lead to success. In the log I NEVER SEE that Gateway's Proxy IP utilized to make the call I do see where it looks like things are trying to move forward [NOTICE] switch_channel.c:1055 New Channel sofia/external-ipv6/nnnnnnnnnn And subsequent log lines continue to reference sofia/external-ipv6/...... Such as: Send signal sofia/external-ipv6/nnnnnnnnnn [BREAK] BUT at the network level my server is only using IPV4 and not utilizing IPV6 at all. And NOT defined to be included in external-ipv6.xml Why is this not utilizing the Gateway defined in external.xml ?

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Mod_callcenter
by Ali Jibran 08 Apr '15

08 Apr '15

I was wondering if there is any way to add queues and agents directly through database and bypass the Callcenter.conf altogether? I tried using ODBC in the core. So I have a MYSQL database for callcenter. I can add agents dynamically. Can I do that with queues somehow?

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channel variables
by Ali Jibran 08 Apr '15

08 Apr '15

Hi. I wanted to know if there was any way to stop a gateway from overriding channel variables? I set effective_caller_id to XXXX but it is changed after the calls makes it way through the gateway. Can I somehow force it to use my variables?

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Changing the voicemail recording menu
by Paul Atreides 08 Apr '15

08 Apr '15

Hi how do I change the menu when the voicemail answers for record a new message? I want it to play the - greeting message - record the message - and then hangup after silence Is is possible to change the menu when the user access its voicemail as well? I found the voicemail_ivr.conf.xml, but I cant find any documentation to it in the wiki Thanks

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Requirements details (1-many video conference)
by Adam Ben-Ayoun 08 Apr '15

08 Apr '15

Hi, Following my discussion with Brian West, here is what we look for: We want to be able to have a similar functionality to mod_conference's video-bridge but also to be able to broadcast video from 1 member to all the other members in the conference. We wrote our own mobile client using Google's WebRTC stack, we are going to use iSAC (we are currently using it) and VP8 for video, hence, we do not need to transcode, just pass through the video. Please let me know if you have any questions! Thanks, Adam

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Re: [Freeswitch-users] WebRTC + Chrome = RTP Socket Error!
by Luiz Fernando Softov 08 Apr '15

08 Apr '15

Brian, I put this patch (https://freeswitch.org/jira/browse/FS-7396) in my source, and this not work, so i cloned again the repository from git git clone https://freeswitch.org/stash/scm/fs/freeswitch.git I format the server, recompile and reinstall freeswitch. FreeSWITCH Version 1.5.15b+git~20150407T231134Z~d55c4a053b~64bit (git d55c4a0 2015-04-07 23:11:34Z 64bit) But this not solve the problem Is the same, it's work fine in firefox, but i get Socket Error! I tried to call from MicroSIP to Chrome - Chrome to MicroSIP Zoiper to Chrome - Chrome to Zoiper Firefox to Chrome - Chrome to Firefox And always is the same when i'm using Chrome (41.0.2272.118) in one leg There is the trace: https://pastebin.freeswitch.org/24100 I'm doing something wrong? -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

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Reply a post in list
by Luiz Fernando Softov 08 Apr '15

08 Apr '15

Am I doing something wrong when I answer a post? Lets say the subject is "Post Test" The title becomes [Freeswitch-users] Post Test ****** When replying, please edit your Subject line so it is more specific ****** than "Re: Contents of FreeSWITCH-users digest..." So, when i reply, i tried Re: [Freeswitch-users] Post Test Re: Post Test re: [Freeswitch-users] Post Test re: Post Test But it's always create a new thread I'm wrong? ps. i'm using gmail -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

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Mod_xml_radius - not able to establish a voicemail call
by Gustavo Silva 08 Apr '15

08 Apr '15

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Feature Comfort Noise to select different codecs on both legs of a Call
by ajinkya Betwar 08 Apr '15

08 Apr '15

Hello Everyone, This is Something New. I am using Freeswitch 1.5 version. I am trying to add comfort noise in media by bridge_generate_comfort_noise variable. But It is choosing same codec for both legs of a call, Instead I want it to select different codecs for both A leg and B leg. In some situation , I need to have 8000 bps on leg A and 16000 bps on leg B, But bridge_generate_comfort_noise by default selects same codec on both leg. How do I select different codecs for comfort noise on both legs ? Please Help in this regard. Thanks. Ajinkya

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suggest best codecs to use with freeswitch for good voice quality ??
by Shabbir abbasi 08 Apr '15

08 Apr '15

for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine suggest best codecs to use with freeswitch and asterisk for good voice quality ??

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