April 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Siproutes incompatible_destination - solved
by Avi Marcus 25 Apr '14

25 Apr '14

If you've been getting incompatible_destination on your siproutes calls, it seems some underlying carrier can no longer handle type 13 from CNG. If you have a gateway file for siproutes, you can simply add: <param name="supress-cng" value="true"/> (It's also accessible in the advanced options in fusionpbx) They also recommend setting <param name="auto-rtp-bugs" value="clear"/> because of sonus, but I can't find on the wiki what that does. -Avi Marcus

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Send users wav of voicemail via sendmail
by Sean Devoy 25 Apr '14

25 Apr '14

Hi All, Can someone explain (or point to documentation) to get voice mail messages emailed to the end user. I have been over and over the mod_voicemail wiki page at https://wiki.freeswitch.org/wiki/Mod_voicemail. It is a nice reference page as it lists lots and lots of voicemail variables and parameters. However, I cannot determine what the minimum rwquirements are. Here is what I have so far: In /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml I have: <param name="mailer-app" value="sendmail"/> <param name="mailer-app-args" value="-t"/> <param name="dump-cores" value="yes"/> In the user directory entry I have: <param name="vm-mailto" value="sdevoy(a)bizfocused.com"/> <param name="vm-mailfrom" value="voicemail(a)bizfocused.com"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true" /> I can send emails from SSH when logged in as root. mail -s "Test mail from testuser1" sdevoy(a)bizfocused.com Hello this is the test mail . I can't find any errors in the console log, but no email gets sent. Have I missed something about creating a freeswitch user in sendmail??? Thanks, Sean

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Bypass media with callback
by FSX 25 Apr '14

25 Apr '14

I'm trying to use GV callback feature in FS and having trouble with making it work in bypass media mode. Incoming calls are working well. Call comes to public context, then it's transferred to local domain context. Then it goes via: <action application="set" data="bypass_media=true"/> <action application="bridge" data="..."/> RTP streams directly between remote server (caller party) and local SIP client, as it should be by design. Calls have better audio quality and don't use recourses of local SIP server. Now, I try to do the same when I use GV callback feature. Local dial plan (as recommended in Google Voice API <https://wiki.freeswitch.org/wiki/Google_Voice_API> ) sets hash with current "uuid" and runs system call to make callback. Then in public context dial plan call is accepted, using "intercept" application with saved "uuid". Exactly as it's recommended. It's working well, but I can't get bypass mode whatever I try to do. * I tried to set "bypass_media=true" before run system call to make callback * I tried to set it before executing "intercept" application * I tried to apply global setting to make sure that FS is always using bypass mode ("inbound-proxy-media" set to "true" in "sip_profiles\internal.xml") Nothing helps so far... FS always ignore those settings and proxies RTP stream. How to make bypass media mode work in this scenario? Thank you in advance -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Bypass-media-with-callback-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Dual WAN router suggestions
by Lloyd Aloysius 25 Apr '14

25 Apr '14

Hello All This is not a freeswitch question. I am sorry. But looking for a Dual WAN router suggestion? I try the pfSense and find a huge limitation, when WAN interface down. It keeps the firewall states and phones never able to connect through the second wan, unless someone clear the firewall rules manually. Any suggestions appreciated Thank you LLoyd

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Oubound ESL: unbridge
by Grant Bagdasarian 24 Apr '14

24 Apr '14

Hello, I'm working on a C# application which takes control over an inbound call in Freeswitch using ESL outbound and it works like a charm! I can answer the call, play something, get digits, etc. Now I'm at the point of implementing bridge functionality in the application. Which one is the best approach and why? 1) From the Outbound ESL Application use the originate command to create a new outbound call and then use uuid_bridge to connect both incoming and outgoing? 2) Use the bridge command from the esl application? I would assume the first option would give me control over the b leg created as well, and the second option would not? This also brings me to my next question: - Is there a way to "unbridge" both call legs, do some stuff on the incoming (a) leg, and then bridge them back without disconnecting any of the legs? Regards, Grant

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How transport sip package via tcp protocol?
by rockxsj 24 Apr '14

24 Apr '14

I want to use tcp to transport the sip package from pjsip or linphone.But I set use tcp on the client,I found it connot connect to the server.Is there some config file should be modify?

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Exporting Diversion Header Twice
by Sanjiv Kumar 24 Apr '14

24 Apr '14

I am trying to export sip_h_Diversion header twice but I only see the last attempt on leg-B. It appears that second export attempt overwrites the first one. How can I export it twice? <extension name="public_did"> <condition field="destination_number" expression="^9999(\d{10})$"> <action application="set" data="ocn=18005551212"/> <action application="set" data="rdn=18005552323"/> <action application="export" data=" nolocal:sip_h_Diversion =<sip:${ocn}@${network_addr}>;reason=user-busy;counter=3;privacy=off"/> <action application="export" data=" nolocal:sip_h_Diversion =<sip:${rdn}@${network_addr}>;reason=no-answer;counter=1;privacy=off"/> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="continue_on_fail=true"/> <action application="transfer" data="1001 XML default"/> </condition> </extension>

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REGISTRATION
by Miguel Oyarzo 24 Apr '14

24 Apr '14

Hi all, I'm sorry for this basic question, I'm new in FS. Is it possible to accept any username&password for REGISTER messages in FS I'm trying to do it in directory/users.xml <user id="*"> but it doesn't work (this file force me to use a username&password for REGISTER). Is there any simple way to accept any user id? Scenario: Client ----> REGISTER (any user/pass) ---> FS Client <--- 200 OK <------------------------- FS Thanks, -- ================================== Miguel Oyarzo DevOps & VoIP Engineer Linux User: # 483188 - counter.li.org http://au.linkedin.com/in/mikeaustralia Melbourne, Australia ==================================

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Distinctive Ring
by Sean Devoy 24 Apr '14

24 Apr '14

Hi All, I want to make the destination phone have a different ring for Internal Calls and External Calls. We use Cisco SPA504Gs which have 10 system and 2 user defined rings to choose from for "default ring". They also have 10 "alternate ring patterns", but I am at a loss how to cause the phone to use an alternate ringer. Everything I see in the wiki has to do with ringback. Unless I misunderstand, that is the ring the caller hears. I want the phone to notify the callee this is a different call type. The only thing I have come up with is to setup 2 different Line Keys with different ringers and direct the calls appropriately. Besides requiring an extra <user>, it complicates the dialplan significantly. Thanks in advance, Sean

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How to dial out via a local extension? (possible?) - OBI110
by Mario G 23 Apr '14

23 Apr '14

Baffled for the last 2 days… I have an OBI110 connected to FreeSwitch as a user, so that it cal dial 911 out via a PSTN. Inbound is fine, but not matter what I do, it only dials out it’s own extension number (8181) on the PTSN line. I ran traces, etc. and don’t see what I am dialing, for example 3511, going to it. I tried to make it a gateway but FS returns 501 not supported. There are various web references to people using it for in/out on Asterisk and FreePBX but so put together FreeSwitch definitions in a similar way. I would really appreciate any suggestions at this point. I have tried many things, nothing seems to work. It always dials 8181 out. Thanks, Mario G <extension name="special.8181"> <condition field="destination_number" expression="^3([23456789]11$)$">  <action application="export" data="destination_number=$1"/-->  <action application="bridge" data="sofia/internal/$1%${sip_profile}"> </condition>

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