Re: [Freeswitch-users] Dialing Out Problem via Gateway
Thank you Brian, The problem is very simple, I or the other party cannot hear each other when I first dial and the other party picks up the phone. We hear the phone ring, the other end picks up the phone says something, but I cannot hear - nothing, even static. Same thing happen on my end, I say something, and the other end do not hear a thing. When I put the call on hold, the other end can hear music on hold. When I take the call back, now we can talk. Something does not go through when the other end picks up the call. This is the extension in the dialplan/default.xml if (typeof YAHOO == "undefined") { var YAHOO = {}; } YAHOO.Shortcuts = YAHOO.Shortcuts || {}; YAHOO.Shortcuts.hasSensitiveText = false; YAHOO.Shortcuts.sensitivityType = []; YAHOO.Shortcuts.doUlt = false; YAHOO.Shortcuts.location = "us"; YAHOO.Shortcuts.document_id = 0; YAHOO.Shortcuts.document_type = ""; YAHOO.Shortcuts.document_title = "t"; YAHOO.Shortcuts.document_publish_date = ""; YAHOO.Shortcuts.document_author = "willbelair@yahoo.com"; YAHOO.Shortcuts.document_url = ""; YAHOO.Shortcuts.document_tags = ""; YAHOO.Shortcuts.document_language = "english"; YAHOO.Shortcuts.annotationSet = { "lw_1232141334_0": { "text": "9054516117", "extended": 0, "startchar": 391, "endchar": 400, "start": 391, "end": 400, "extendedFrom": "", "predictedCategory": "", "predictionProbability": "0", "weight": 1, "relScore": 0, "type": ["shortcuts:/us/instance/identifier/fedex_tracking"], "category": ["IDENTIFIER"], "wikiId": "", "relatedWikiIds": [], "relatedEntities": [], "showOnClick": [], "context": "", "metaData": { "verified": "false", "visible": "true" } } }; YAHOO.Shortcuts.headerID = "a9059b1f35336b4363b0c75035b61d07"; <extension name="mygateway"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/mygw/$1"/> </condition> </extension> --- On Fri, 1/16/09, Brian West <brian@freeswitch.org> wrote: From: Brian West <brian@freeswitch.org> Subject: Re: [Freeswitch-users] Dialing Out Problem via Gateway To: freeswitch-users@lists.freeswitch.org Date: Friday, January 16, 2009, 1:13 PM Can you detail your problem a bit more? /b On Jan 16, 2009, at 3:09 PM, Will Smith wrote: Hi, I got a strange problem that I don't really understand, and I hope that you could give me some hint how to fix that: When I dial out through a gateway that is defined in the sip_profiles/external , (The xml file is simple as below. ) I cannot talk or hear from the other end. But when I put the line on hold, two ends can hear music, and when open the line again, this time 2 ends can hear and talk. Is there any where that I can fix this problem? Thank you <include> <gateway name="mygw"> <param name="username" value="myusername"/> <param name="realm" value="my_sip_server.com"/> <param name="password" value="mypassword"/> <param name="proxy" value="my_sip_server.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> </gateway> </include> _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
NAT involved? /b On Jan 16, 2009, at 3:30 PM, Will Smith wrote:
Thank you Brian,
The problem is very simple, I or the other party cannot hear each other when I first dial and the other party picks up the phone. We hear the phone ring, the other end picks up the phone says something, but I cannot hear - nothing, even static. Same thing happen on my end, I say something, and the other end do not hear a thing. When I put the call on hold, the other end can hear music on hold. When I take the call back, now we can talk. Something does not go through when the other end picks up the call. This is the extension in the dialplan/default.xml
<extension name="mygateway"> <condition field="destination_number" expression="^(1{0,1}\d{10}) $"> <action application="set" data="effective_caller_id_number=12223334444"/> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/mygw/$1"/> </condition> </extension>
Well, if NAT involved, why did I get through after I put the call on hold and take the call back. I am getting the SIP trace, hope that will show something. Thank you all --- On Fri, 1/16/09, Brian West <brian@freeswitch.org> wrote: From: Brian West <brian@freeswitch.org> Subject: Re: [Freeswitch-users] Dialing Out Problem via Gateway To: freeswitch-users@lists.freeswitch.org Date: Friday, January 16, 2009, 1:41 PM NAT involved? /b On Jan 16, 2009, at 3:30 PM, Will Smith wrote: Thank you Brian, The problem is very simple, I or the other party cannot hear each other when I first dial and the other party picks up the phone. We hear the phone ring, the other end picks up the phone says something, but I cannot hear - nothing, even static. Same thing happen on my end, I say something, and the other end do not hear a thing. When I put the call on hold, the other end can hear music on hold. When I take the call back, now we can talk. Something does not go through when the other end picks up the call. This is the extension in the dialplan/default.xml <extension name="mygateway"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/> <!-- If your provider does not provide ringback (180 or 183) you may simulate ringback by uncommenting the following line. --> <!-- action application="ringback" /--> <action application="bridge" data="sofia/gateway/mygw/$1"/> </condition> </extension> _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I would suspect the NAT wasn't punching holes or lied. :) /b On Jan 16, 2009, at 3:57 PM, Will Smith wrote:
Well, if NAT involved, why did I get through after I put the call on hold and take the call back. I am getting the SIP trace, hope that will show something. Thank you all
participants (2)
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Brian West
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Will Smith