dear all i'm new to freeswitch, and i found difficulties to connect fs to my mera switch. here's the topology extension-------------->FS-------------> meraswitch----------------->pstn what need to be configured to connect FS to meraswitch that run SIP protocol TIA budi wibowo
It depends on what version of Mera that you are running. The older version, you need to have SIPHIT running; however, the newer one like Mera Pro or Mera II, they should connect with no problem. -djbinter On Thu, May 6, 2010 at 4:42 PM, budi wibowo <bwibowo@gmail.com> wrote:
dear all i'm new to freeswitch, and i found difficulties to connect fs to my mera switch. here's the topology
extension-------------->FS-------------> meraswitch----------------->pstn
what need to be configured to connect FS to meraswitch that run SIP protocol
TIA
budi wibowo
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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 Tia Budi -----Original Message----- From: DJB INTERNATIONAL <djbinter@gmail.com> Date: Thu, 6 May 2010 20:34:23 To: <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] outgoing gateway _______________________________________________ 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
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
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 <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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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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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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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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That won't help if you don't put a full log and your conf on pastebin. 2010/5/8 budi wibowo <bwibowo@gmail.com>:
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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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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hi, I am looking into event_socket and the C# demo in contributions, but I get a request for a password when I try to use this? EventSocket.pfx Jan _________________________________________________________________ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969
Hi, check libs\esl\managed Diego Toro http://lacarretade.blogspot.com/ --- On Sat, 5/8/10, Jan Berger <janvb@live.com> wrote: From: Jan Berger <janvb@live.com> Subject: [Freeswitch-users] C#/EventSocket To: freeswitch-users@lists.freeswitch.org Date: Saturday, May 8, 2010, 5:51 AM hi, I am looking into event_socket and the C# demo in contributions, but I get a request for a password when I try to use this? EventSocket.pfx Jan Hotmail: Free, trusted and rich email service. Get it now. -----Inline Attachment Follows----- _______________________________________________ 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
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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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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You need to add this in your gateway config: <param name="username" value="none"/> <param name="password" value="none"/> and change the register param to: <param name="register" value="false"/> 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 09/05/2010 à 02:16, budi wibowo a écrit :
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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thx a lot, refering to http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start with required changes but still failed, i also attached the F8 capture, hope can solve this problem regards budi <http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start> On Sun, May 9, 2010 at 4:49 PM, David Ponzone <david.ponzone@gmail.com>wrote:
You need to add this in your gateway config:
<param name="username" value="none"/> <param name="password" value="none"/>
and change the register param to: <param name="register" value="false"/>
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 09/05/2010 à 02:16, budi wibowo a écrit :
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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-- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn
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dear all thx for your support it's working now :) next question is after i change some xml file then i do reloadxml, but in fact not all config changes. reason of my statement is: many time i do reload xml but seems my config still not work, then i shutdown the FS, then invoke ./freeswitch again and work any advice? regards budi On Mon, May 10, 2010 at 5:06 AM, budi wibowo <bwibowo@gmail.com> wrote:
thx a lot, refering to http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start with required changes but still failed, i also attached the F8 capture, hope can solve this problem
regards budi
<http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>
On Sun, May 9, 2010 at 4:49 PM, David Ponzone <david.ponzone@gmail.com>wrote:
You need to add this in your gateway config:
<param name="username" value="none"/> <param name="password" value="none"/>
and change the register param to: <param name="register" value="false"/>
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 09/05/2010 à 02:16, budi wibowo a écrit :
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 > > > > _______________________________________________ > 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 >
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budi wibowo <bwibowo@gmail.com> wrote:
dear all thx for your support it's working now :) next question is after i change some xml file then i do reloadxml, but in fact not all config changes.
It isn't supposed to.
reason of my statement is: many time i do reload xml but seems my config still not work, then i shutdown the FS, then invoke ./freeswitch again and work any advice?
If you're changing the SIP configuration, try sofia profile <profile-name> rescan reloadxml or if that isn't enough, sofia profile <profile-name> restart reloadxml Reloadxml just loads the new configuration files; it does *not* re-initialize all of the configuration parameters.
Thx again, now I move to cdr part. Both cdr in csv and xml are working, but I want to store in mysql db. Db structure already created, following the wiki doc for mod_cdr. But I don't see the cdr logged into mysql, pressing F8 also not showing any db query. Any help? Mod_cdr loaded via modules.conf.xml TIA Budi wibowo -----Original Message----- From: Jason White <jason@jasonjgw.net> Date: Mon, 10 May 2010 10:18:16 To: <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] outgoing gateway budi wibowo <bwibowo@gmail.com> wrote:
dear all thx for your support it's working now :) next question is after i change some xml file then i do reloadxml, but in fact not all config changes.
It isn't supposed to.
reason of my statement is: many time i do reload xml but seems my config still not work, then i shutdown the FS, then invoke ./freeswitch again and work any advice?
If you're changing the SIP configuration, try sofia profile <profile-name> rescan reloadxml or if that isn't enough, sofia profile <profile-name> restart reloadxml Reloadxml just loads the new configuration files; it does *not* re-initialize all of the configuration parameters. _______________________________________________ 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
On Mon, May 10, 2010 at 5:19 AM, Budi wibowo <bwibowo@gmail.com> wrote:
Thx again, now I move to cdr part. Both cdr in csv and xml are working, but I want to store in mysql db. Db structure already created, following the wiki doc for mod_cdr. But I don't see the cdr logged into mysql, pressing F8 also not showing any db query. Any help? Mod_cdr loaded via modules.conf.xml TIA Budi wibowo
mod_cdr has long since been deprecated. You need to use mod_cdr_csv to write out the SQL statements and use a cron job to insert them into your database. Check out this page: http://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_SQL_Script_to_create_MyS... There are some examples of how people are doing this. -MC
Thx a lot michael -----Original Message----- From: Michael Collins <msc@freeswitch.org> Date: Mon, 10 May 2010 15:57:43 To: <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] outgoing gateway _______________________________________________ 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
participants (10)
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budi wibowo
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Budi wibowo
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David Ponzone
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Diego Toro
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DJB INTERNATIONAL
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Jan Berger
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Jason White
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Michael Collins
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Michael Jerris
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Seven Du