again, pastebin your freeswitch logs so others can help you. don't forget to press F8 before capture your logs. 2010/5/9 budi wibowo <bwibowo@gmail.com>:
thx a lot for your explanation, i have another question, my sip server (mera sip-hit) dont need user name and password to connect. for security i just use firewall rules. if i refer to other samples like fwd,voicheap etc all require user name and password. my conf/sip_profiles/external/test.xml look like this <include> <gateway name="test"> <param name="realm" value="202.xx.xx.xx.xx"/> <param name="proxy" value="202.xx.xx.xx.xx"/> <param name="expire-seconds" value="3600"/> <param name="register" value="true"/> <param name="retry-seconds" value="3600"/> </gateway> </include>
my conf/dialplan/default/00_test.xml look like <include> <extension name="test"> <condition field="destination_number" expression="^(\d)$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/test/$1"/> </condition> </extension> </include>
but still i cant see any call coming from my mera siphit. TIA budi wibowo
On Sat, May 8, 2010 at 4:04 PM, David Ponzone <david.ponzone@gmail.com> wrote:
Not sure because your logs are toon short, but it sems that your call is routed to ENUM. In the default config, ENUM routing is the last resort rule. You need to add your own stuff before it. To do so, you may add your extensions in a xml file called 01_something.xml in conf/dialplan/default (see 01_example.com.xml for an example). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 08/05/2010 à 10:26, budi wibowo a écrit :
dear all i try to make some changes but my call still failing why the call never reach gateway i define in sip_profile/external/ and call goes to voiprakyar.or.id that i never define
2010-05-08 15:22:46.403733 [NOTICE] switch_channel.c:602 New Channel sofia/internal/budi@sip1.xxx.com [7e483bd8-ba9d-4f1e-8560-74edb8fba2af] 2010-05-08 15:22:46.407056 [INFO] mod_dialplan_xml.c:315 Processing budi->62815145150 in context default 2010-05-08 15:22:46.410565 [NOTICE] switch_ivr.c:1349 Transfer sofia/internal/budi@sip1.xxx.com to enum[62815145150@default] 2010-05-08 15:22:46.750534 [NOTICE] switch_channel.c:602 New Channel sofia/internal/62815145150@voiprakyat.or.id [1e55a5c7-5d58-4437-a48b-4278154b57a0] 2010-05-08 15:22:48.426621 [NOTICE] sofia.c:3849 Hangup sofia/internal/62815145150@voiprakyat.or.id [CS_CONSUME_MEDIA] [CALL_REJECTED]
TIA budi On Sat, May 8, 2010 at 9:30 AM, Seven Du <dujinfang@gmail.com> wrote:
In the mean time, I think it's a good article for you:
http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
2010/5/8 Seven Du <dujinfang@gmail.com>:
default configuration files of mod_sofia is in conf/sip_profiles
http://wiki.freeswitch.org/wiki/Sofia http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
2010/5/8 budi wibowo <bwibowo@gmail.com>:
thx, after reading the document i still got confusion about sofia module i give example below
Dialing Through A Gateway(SIP Provider)
A gateway is a means for making outbound calls through a SIP provider. For example:
sofia/gateway/mygateway.com/1234
by default i dont find any directory named sofia, where i should put this directory? many document i read telling about sofia TIA budi
On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike@jerris.com> wrote:
http://wiki.freeswitch.org/wiki/Dialplan On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
Yes I have siphit installed, I tried to make some changes on dialplan file but call always goes to other server, what should I configure to implement this outgoing call
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