February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSwtich DB
by Joe Flemmings 06 Feb '12

06 Feb '12

Is their a reason why freeswitch still creates DB files in /usr/local/freeswitch/db/ even when using ODBC at the core as decsribed here http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

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No outbound audio on new fs build
by Ira Tessler 06 Feb '12

06 Feb '12

The following are two pcap files that have one outbound call in them. Everything is identical except the version of Freeswitch. With the later version of Freeswitch, I am getting one way audio on the outbound call. On the later version of Freeswitch, there is an Update sip message that is not in the older version. Can anyone help me and let me know how I can fix the one way audio? I only get the one way audio when using Yealink phones behind an Edgemarc router. Old version of Freeswitch pcap: http://dl.dropbox.com/u/47191639/lan_side_yealink_t28_good_audio_edgemarc_o… New Version of Freeswitch pcap: http://dl.dropbox.com/u/47191639/lan_side_yealink_t28_no_audio_edgemarc.pcap Thanks, Ira Tessler ConnectMe (732) 490-9007 x2 www.connectmevoice.com

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Re: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ?
by Bob Smith 06 Feb '12

06 Feb '12

Welcome back by the way Brian... ;-) If only it were that easy...... in conf/directory/default.xml I changed.... <param name="dial-string" value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> to read <param name="dial-string" value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain}),sofia/other_profile/${dialed_user}}"/> Then the OpenVPN phone rings, but the internally registered phone does not ? > You'll have to modify the dial-string for the user 1000 to make sure it searches this new profile ...

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One Way Audio, from one side.
by Tariq Mahmood 06 Feb '12

06 Feb '12

Hi, I hope you guys are doing great. I am having an issue with my number, When i dial the number with my asterisk server, the callee can hear me but i cant hear him. But if the same guy call me, the conversation goes great. Any Help??? -- Best Regards Tareq Khan

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Re: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ?
by Bob Smith 06 Feb '12

06 Feb '12

In the public dial plan…… ============================== <extension name="mytest"> <condition field="destination_number" expression="^(4412345678)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="1000 XML default"/> </condition> </extension> "1000" being a user under directory/default/ =========================================== <include> <user id="1000"> <params> <param name="a1-hash" value="ffffffffffffffffffff"/> <param name="vm-disk-quota" value="1500"/> <param name="vm-a1-hash" value="ffffffffffffffffffff"/> </params> <variables> <variable name="toll_allow" value="uk_nopremium,int_standard,int_usaexpremium"/> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Some User"/> <variable name="effective_caller_id_number" value="00000000"/> --> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="mycallgroup"/> </variables> </user> </include> >You're getting it because it can't find the USER to call. How are you calling the users on this interface in your dialplan?

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Custom filename - XML CDR
by Carlo Dimaggio 06 Feb '12

06 Feb '12

Hi Guys, I need a prefix in the filename created my mod_xml_cdr (ex: <accountcode or other var>-<uuid>.cdr.xml). Is there any way to set this custom filename in the dialplan? A reason could be performance in the billing procedure in order to process only that particular account and not the entire directory. Best regards,

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Getting USER_NOT_REGISTERED for no reason ?
by Bob Smith 06 Feb '12

06 Feb '12

I am confused.... ;-( Basically, the only reason the openvpn profile exists is to enable FreeSwitch to listen on the local OpenVPN IP. Other than that, I want OpenVPN profile users to be treated the same way as Internal profile users ? In order to achieve that goal, do I need to do more than just copy internal.xml to a new file and change the profile name and IP addresses ? > You are using the internal profile when your phone is clearly registered on the openvpn profile.

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event_socket.conf
by Miha Zoubek 06 Feb '12

06 Feb '12

Hi, I have changed default password for event_socket.conf (clueclone). Now I am unable to connect to FS via fs_cli (I can connect remotely). I have also set ACL which will also do it for localhost. Where can I find configuration file for FS_cli that I can change default password (If I set password to clueclone in event_socket.conf it works). I have try to change it in: /usr/local/src/freeswitch/libs/esl/fs_cli, but I guess this is not the right file. My ACL: <list name="php" default="deny"> <node type="allow" cidr="xxx.xxx.xxx.xxx/29"/> <node type="allow" cidr="127.0.0.0/30"/> </list> Thanks! Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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Getting USER_NOT_REGISTERED for no reason ?
by Bob Smith 06 Feb '12

06 Feb '12

As you can see from http://pastebin.freeswitch.org/18299, I have a Snom handset successfully registered with my FreeSwitch instance over an OpenVPN connection. As per the trace, I can make outbound calls without issue. But inbound, FreeSwithc seems to think I am not registered when I am ? To make absolutley sure I am registered, I have issued a " sofia profile <profile_name> flush_inbound_reg reboot" and sure enough, the Snom unit reboots and re-registeres without issue. Help ! Thanks Bob

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Striping destination number
by Miha Zoubek 06 Feb '12

06 Feb '12

Hi, how can I strip destination number in my dialplan if the destination number is not in condition option. <anti-action application="bridge" data="sofia/external/386${sip_redirect_contact_user_0}(a)xxx.xxx.xxx.xxx" /> I need to stip leading zero from ${sip_redirect_contact_user_0} variable. Thanks! Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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