VoIP Mailing List Archives

[brian at freeswitch.org]

Can you detail your problem a bit more?

/b

On Jan 16, 2009, at 3:09 PM, Will Smith wrote:

[willbelair at yahoo.com]

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>

[willbelair at yahoo.com]

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:

[msc at freeswitch.org]

A SIP trace would be extremely helpful.
http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch#Enabling_SIP.2FSofia_Tracing
-MC

On Fri, Jan 16, 2009 at 1:13 PM, Brian West <brian@freeswitch.org> wrote:

[brian at freeswitch.org]

NAT involved?

/b

On Jan 16, 2009, at 3:30 PM, Will Smith wrote:

[willbelair at yahoo.com]

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:

[brian at 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:

[cyberalby at gmail.com]

Hi,
I've the same problem.
I've create the log with only the problem (the log is in attach).
One gateway (messagenet.it) and two internal account (1002 and 1008).
My dialplan:
<include>
<extension name="messagenet.it">
<condition field="destination_number" expression="^(5300000)$">
<action application="set" data="domain_name=$${domain}"/>
<action application="transfer" data="1008 XML default"/>
</condition>
</extension>
</include>

My Sip Profile:
<include>
<gateway name="messagenet.it">
<param name="username" value="5300000"/>
<param name="realm" value="sip.messagenet.it"/>
<param name="from-domain" value="sip.messagenet.it"/>
<param name="password" value="password"/>
<param name="register" value="true"/>
<param name="retry-seconds" value="30"/>
</gateway>
</include>

My freeswith is runnig on Windows Server 2003 with 2 NIC (1 interna 192.168.0.15 and 1 external 10.0.3.6 (212.001.001.001))

For privacy I've change my IP and my phone number.

--
Alby

[cyberalby at gmail.com]

[brian at freeswitch.org]

You modified the Local_Extension in default.xml

2009-01-17 16:10:08 [DEBUG] mod_dialplan_xml.c:117 parse_exten()
Regex: [Local_Extension] destination_number(1008) =~ /^+39020000000$/
2009-01-17 16:10:08 [ERR] switch_regex.c:94 switch_regex_perform()
COMPILE ERROR: 1 [nothing to repeat][^+39020000000$]
2009-01-17 16:10:08 [DEBUG] mod_dialplan_xml.c:119 parse_exten() Regex
mismatch


Remove that match for the /^+39020000000$/ and put it back to the
default value... it should work fine.

/b

On Jan 17, 2009, at 10:02 AM, Alberto Ceccarelli wrote:

[cyberalby at gmail.com]

Sorry, but the problem remains.
I can add some details.
After response, we can listen the DTMF tones, but not the voice.

Now my conf. files:

vars.xml:
<X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
<X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>


dialplan\default.xml:
<extension name="Local_Extension">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="dialed_extension=$1"/>
</condition>
<condition field="destination_number" expression="^${caller_id_number}$">
<action application="set" data="voicemail_authorized=${sip_authorized}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="check default ${domain_name} ${dialed_extension}"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> -->
<anti-action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
<anti-action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<anti-action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
<anti-action application="set" data="ringback=${us-ring}"/>
<anti-action application="set" data="transfer_ringback=$${hold_music}"/>
<anti-action application="set" data="call_timeout=30"/>
<!-- <anti-action application="set" data="sip_exclude_contact=${network_addr}"/> -->
<anti-action application="set" data="hangup_after_bridge=true"/>
<!--<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> -->
<anti-action application="set" data="continue_on_fail=true"/>
<anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
<anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
<!--<anti-action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
<anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
<anti-action application="answer"/>
<anti-action application="sleep" data="1000"/>
<anti-action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
</condition>
</extension>
Ciao.
Alby.


2009/1/19 Brian West <brian@freeswitch.org (brian@freeswitch.org)>