Some Newbie questions about dialplan and local Sip registration
Hi List, for the first experiments with freeswitch I downloaded the Windows installation. Now Im trying to get my 2 Sipphones get connected to. Later I want connect the freeswitch to my asterisk gateway. I find the examples pretty complex therfore Im trying to build up a simple solution to understand the functions from the scratch .. my current problem is , that I cant route my local sips to each other ( registration seems to work now). the next is , that freeshwitch is not able to connect to asterisk. but I will describe this later. I installed in the Directory a xml file ( called 22.xml) with the following content : <include> <domain name="$${domain}"> <user id="22" mailbox="22"> <params> <param name="password" value="Xk21%"></param> <param name="vm-password" value="22"></param> <param name="sip-port" value="5060"></param> </params> <variables> <variable name="accountcode" value="22"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 22"></variable> <variable name="effective_caller_id_number" value="22"></variable> </variables> </user> <user id="24" mailbox="24"> <params> <param name="password" value="dudeldum"></param> <param name="vm-password" value="24"></param> <param name="sip-port" value="5060"></param> </params> <variables> <variable name="accountcode" value="24"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 24"></variable> <variable name="effective_caller_id_number" value="24"></variable> </variables> </user> </domain> </include> This seems to be ok now. Now I want to dial from 22 to 24 , wherefore I configured this dialplan : <include> <context name="any"> <condition field="destination_number" expression="^(2[0-9])$"> <action application="bridge" data="user/${dialed_extension}"/> </condition> </include> wich doesnt work , mybe b/c the user/${dialed_extension} I dont know... Freeswitch says: [INFO] switch_core_state_machine.c:136 No Route, Aborting [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/24@192.168.1.34 [CS_ROUTING] [NO_ROUTE_DESTINATION] [NOTICE] switch_core_session.c:1086 Session 17 (sofia/internal/24@192.168.1.34) Ended [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/24@192.168.1.34 [CS_DESTROY] Im sure , for you guys this cant be a big deal;) Next Point is my Asterisk registration , mybe you can help me out here to .. : In the sip-profiles/external I installed the my_asterisk.xml with that content : <include> <gateway name="asterisk"> <param name="username" value="28"></param> <param name="password" value="test"></param> <param name="realm" value="28"></param> <param name="proxy" value="192.168.1.119"></param> <param name="register" value="true"></param> </gateway> </include> Freeswitch allways complains a timeout error : [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request Timeout [408]. failure #17 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry to 540 seconds. it seems that It cant connect ( I also tried out to explicit set the port to 5060 b/c I read something about 5080 .. : <param name="sip-port" value="5060"></param> but this didnt help) In my Asterisk I set in the sip.conf the entry 28 with the pw test .... If someone could help me with my first steps I would be verrry thankful ;)) cheers Filip -- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis
hi Filip, for calling a user... please read this first: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered... for making a GW register into e.g. asterisk please use this: <include> <gateway name="gw01"> <param name="username" value="USERNAME_ON_ASTERISK"/> <param name="realm" value="ASTERISK_IP_ADDRESS"/> <param name="password" value="PASSWORD_ON_ASTERISK"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> </gateway> </include> this should be enough to register the GW... after that please read this: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gate... in your case it will be something like this: <extension name="dialGW"> <condition field="destination_number" expression="^(NUMBER_TO_SEND_TO_ASTERISK)$"> <action application="bridge" data="sofia/gateway/gw01/$1"/> </condition> </extension> On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker <lyncker@lyth.de> wrote:
Hi List,
for the first experiments with freeswitch I downloaded the Windows installation. Now Im trying to get my 2 Sipphones get connected to. Later I want connect the freeswitch to my asterisk gateway.
I find the examples pretty complex therfore Im trying to build up a simple solution to understand the functions from the scratch ..
my current problem is , that I cant route my local sips to each other ( registration seems to work now). the next is , that freeshwitch is not able to connect to asterisk. but I will describe this later.
I installed in the Directory a xml file ( called 22.xml) with the following content :
<include> <domain name="$${domain}"> <user id="22" mailbox="22"> <params> <param name="password" value="Xk21%"></param> <param name="vm-password" value="22"></param> <param name="sip-port" value="5060"></param>
</params> <variables> <variable name="accountcode" value="22"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 22"></variable> <variable name="effective_caller_id_number" value="22"></variable> </variables> </user> <user id="24" mailbox="24"> <params> <param name="password" value="dudeldum"></param> <param name="vm-password" value="24"></param> <param name="sip-port" value="5060"></param>
</params> <variables> <variable name="accountcode" value="24"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 24"></variable> <variable name="effective_caller_id_number" value="24"></variable> </variables> </user> </domain> </include>
This seems to be ok now. Now I want to dial from 22 to 24 , wherefore I configured this dialplan :
<include> <context name="any"> <condition field="destination_number" expression="^(2[0-9])$">
<action application="bridge" data="user/${dialed_extension}"/>
</condition> </include>
wich doesnt work , mybe b/c the user/${dialed_extension} I dont know... Freeswitch says: [INFO] switch_core_state_machine.c:136 No Route, Aborting [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/24@192.168.1.34 [CS_ROUTING] [NO_ROUTE_DESTINATION] [NOTICE] switch_core_session.c:1086 Session 17 (sofia/internal/24@192.168.1.34) Ended [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/24@192.168.1.34 [CS_DESTROY]
Im sure , for you guys this cant be a big deal;)
Next Point is my Asterisk registration , mybe you can help me out here to .. :
In the sip-profiles/external I installed the my_asterisk.xml with that content :
<include> <gateway name="asterisk"> <param name="username" value="28"></param> <param name="password" value="test"></param> <param name="realm" value="28"></param> <param name="proxy" value="192.168.1.119"></param> <param name="register" value="true"></param> </gateway> </include>
Freeswitch allways complains a timeout error : [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request Timeout [408]. failure #17 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry to 540 seconds.
it seems that It cant connect ( I also tried out to explicit set the port to 5060 b/c I read something about 5080 .. : <param name="sip-port" value="5060"></param> but this didnt help) In my Asterisk I set in the sip.conf the entry 28 with the pw test ....
If someone could help me with my first steps I would be verrry thankful ;))
cheers
Filip
-- _________________________________ Filip Lyncker, Dipl.-Inform. (FH)
Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany
Fon +49 611/9006951 Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399
Geschäftsführer: Filip Lyncker, Armin Theis
_______________________________________________ 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
Hi Tihomir, Thanks for your help , I added the Asteriskparameters as you described below, but I still get the same timeout error: 2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry to 270 seconds. 2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request Timeout [408]. failure #9 Now, my gateway entry looks like the following : <include> <gateway name="asterisk"> <param name="username" value="28"/> <param name="realm" value="192.168.1.119"/> <param name="proxy" value="192.168.1.119"/> <param name="password" value="test"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="sip-port" value="5060"></param> </gateway> </include> What can be still wrong here? Regards, Filip Tihomir Culjaga schrieb:
hi Filip,
for calling a user... please read this first: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered... for making a GW register into e.g. asterisk please use this:
<include> <gateway name="gw01"> <param name="username" value="USERNAME_ON_ASTERISK"/> <param name="realm" value="ASTERISK_IP_ADDRESS"/> <param name="password" value="PASSWORD_ON_ASTERISK"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> </gateway> </include>
this should be enough to register the GW... after that please read this: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gate...
in your case it will be something like this:
<extension name="dialGW"> <condition field="destination_number" expression="^(NUMBER_TO_SEND_TO_ASTERISK)$"> <action application="bridge" data="sofia/gateway/gw01/$1"/> </condition> </extension>
On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker <lyncker@lyth.de <mailto:lyncker@lyth.de>> wrote:
Hi List,
for the first experiments with freeswitch I downloaded the Windows installation. Now Im trying to get my 2 Sipphones get connected to. Later I want connect the freeswitch to my asterisk gateway.
I find the examples pretty complex therfore Im trying to build up a simple solution to understand the functions from the scratch ..
my current problem is , that I cant route my local sips to each other ( registration seems to work now). the next is , that freeshwitch is not able to connect to asterisk. but I will describe this later.
I installed in the Directory a xml file ( called 22.xml) with the following content :
<include> <domain name="$${domain}"> <user id="22" mailbox="22"> <params> <param name="password" value="Xk21%"></param> <param name="vm-password" value="22"></param> <param name="sip-port" value="5060"></param>
</params> <variables> <variable name="accountcode" value="22"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 22"></variable> <variable name="effective_caller_id_number" value="22"></variable> </variables> </user> <user id="24" mailbox="24"> <params> <param name="password" value="dudeldum"></param> <param name="vm-password" value="24"></param> <param name="sip-port" value="5060"></param>
</params> <variables> <variable name="accountcode" value="24"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 24"></variable> <variable name="effective_caller_id_number" value="24"></variable> </variables> </user> </domain> </include>
This seems to be ok now. Now I want to dial from 22 to 24 , wherefore I configured this dialplan :
<include> <context name="any"> <condition field="destination_number" expression="^(2[0-9])$">
<action application="bridge" data="user/${dialed_extension}"/>
</condition> </include>
wich doesnt work , mybe b/c the user/${dialed_extension} I dont know... Freeswitch says: [INFO] switch_core_state_machine.c:136 No Route, Aborting [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34> [CS_ROUTING] [NO_ROUTE_DESTINATION] [NOTICE] switch_core_session.c:1086 Session 17 (sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34>) Ended [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34> [CS_DESTROY]
Im sure , for you guys this cant be a big deal;)
Next Point is my Asterisk registration , mybe you can help me out here to .. :
In the sip-profiles/external I installed the my_asterisk.xml with that content :
<include> <gateway name="asterisk"> <param name="username" value="28"></param> <param name="password" value="test"></param> <param name="realm" value="28"></param> <param name="proxy" value="192.168.1.119"></param> <param name="register" value="true"></param> </gateway> </include>
Freeswitch allways complains a timeout error : [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request Timeout [408]. failure #17 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry to 540 seconds.
it seems that It cant connect ( I also tried out to explicit set the port to 5060 b/c I read something about 5080 .. : <param name="sip-port" value="5060"></param> but this didnt help) In my Asterisk I set in the sip.conf the entry 28 with the pw test ....
If someone could help me with my first steps I would be verrry thankful ;))
cheers
Filip
-- _________________________________ Filip Lyncker, Dipl.-Inform. (FH)
Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany
Fon +49 611/9006951 Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399
Geschäftsführer: Filip Lyncker, Armin Theis
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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
------------------------------------------------------------------------
_______________________________________________ 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
-- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis
hmmm .. can you register using x-lite or some other softphone with the same credentials? can you paste a siptrace of the failed registration? BTW: Make sure nothing is already registered with this credentials when you try with FS T. On Tue, Sep 22, 2009 at 12:56 PM, Filip Lyncker <lyncker@lyth.de> wrote:
Hi Tihomir,
Thanks for your help , I added the Asteriskparameters as you described below, but I still get the same timeout error: 2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry to 270 seconds. 2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request Timeout [408]. failure #9
Now, my gateway entry looks like the following :
<include> <gateway name="asterisk"> <param name="username" value="28"/> <param name="realm" value="192.168.1.119"/> <param name="proxy" value="192.168.1.119"/> <param name="password" value="test"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="sip-port" value="5060"></param> </gateway> </include>
What can be still wrong here?
Regards,
Filip
Tihomir Culjaga schrieb:
hi Filip,
for calling a user... please read this first:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered...
for making a GW register into e.g. asterisk please use this:
<include> <gateway name="gw01"> <param name="username" value="USERNAME_ON_ASTERISK"/> <param name="realm" value="ASTERISK_IP_ADDRESS"/> <param name="password" value="PASSWORD_ON_ASTERISK"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> </gateway> </include>
this should be enough to register the GW... after that please read this:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gate...
in your case it will be something like this:
<extension name="dialGW"> <condition field="destination_number" expression="^(NUMBER_TO_SEND_TO_ASTERISK)$"> <action application="bridge" data="sofia/gateway/gw01/$1"/> </condition> </extension>
On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker <lyncker@lyth.de <mailto:lyncker@lyth.de>> wrote:
Hi List,
for the first experiments with freeswitch I downloaded the Windows installation. Now Im trying to get my 2 Sipphones get connected to. Later I want connect the freeswitch to my asterisk gateway.
I find the examples pretty complex therfore Im trying to build up a simple solution to understand the functions from the scratch ..
my current problem is , that I cant route my local sips to each other ( registration seems to work now). the next is , that freeshwitch is not able to connect to asterisk. but I will describe this later.
I installed in the Directory a xml file ( called 22.xml) with the following content :
<include> <domain name="$${domain}"> <user id="22" mailbox="22"> <params> <param name="password" value="Xk21%"></param> <param name="vm-password" value="22"></param> <param name="sip-port" value="5060"></param>
</params> <variables> <variable name="accountcode" value="22"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 22"></variable> <variable name="effective_caller_id_number" value="22"></variable> </variables> </user> <user id="24" mailbox="24"> <params> <param name="password" value="dudeldum"></param> <param name="vm-password" value="24"></param> <param name="sip-port" value="5060"></param>
</params> <variables> <variable name="accountcode" value="24"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 24"></variable> <variable name="effective_caller_id_number" value="24"></variable> </variables> </user> </domain> </include>
This seems to be ok now. Now I want to dial from 22 to 24 , wherefore I configured this dialplan :
<include> <context name="any"> <condition field="destination_number" expression="^(2[0-9])$">
<action application="bridge" data="user/${dialed_extension}"/>
</condition> </include>
wich doesnt work , mybe b/c the user/${dialed_extension} I dont know... Freeswitch says: [INFO] switch_core_state_machine.c:136 No Route, Aborting [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34> [CS_ROUTING] [NO_ROUTE_DESTINATION] [NOTICE] switch_core_session.c:1086 Session 17 (sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34>) Ended [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34> [CS_DESTROY]
Im sure , for you guys this cant be a big deal;)
Next Point is my Asterisk registration , mybe you can help me out
here
to .. :
In the sip-profiles/external I installed the my_asterisk.xml with
that
content :
<include> <gateway name="asterisk"> <param name="username" value="28"></param> <param name="password" value="test"></param> <param name="realm" value="28"></param> <param name="proxy" value="192.168.1.119"></param> <param name="register" value="true"></param> </gateway> </include>
Freeswitch allways complains a timeout error : [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request Timeout [408]. failure #17 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting
retry
to 540 seconds.
it seems that It cant connect ( I also tried out to explicit set the port to 5060 b/c I read something about 5080 .. : <param name="sip-port" value="5060"></param> but this didnt help) In my Asterisk I set in the sip.conf the entry 28 with the pw test ....
If someone could help me with my first steps I would be verrry thankful ;))
cheers
Filip
-- _________________________________ Filip Lyncker, Dipl.-Inform. (FH)
Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany
Fon +49 611/9006951 Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399
Geschäftsführer: Filip Lyncker, Armin Theis
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
------------------------------------------------------------------------
_______________________________________________ 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
-- _________________________________ Filip Lyncker, Dipl.-Inform. (FH)
Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany
Fon +49 611/9006951 Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399
Geschäftsführer: Filip Lyncker, Armin Theis
_______________________________________________ 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
now i registered from my x-lite client without anyproblems. but I think i got it now, my tcpdump says the following : IP 192.168.1.119.5060 > 93.210.212.xxx.5080: SIP, length: 465 wich is the external IP of my network ! must have somthing todo with NAT / Masquerade options... how can I avoid this ? thanks for your help ... regards, filip Tihomir Culjaga schrieb:
hmmm .. can you register using x-lite or some other softphone with the same credentials?
can you paste a siptrace of the failed registration?
BTW: Make sure nothing is already registered with this credentials when you try with FS
T.
On Tue, Sep 22, 2009 at 12:56 PM, Filip Lyncker <lyncker@lyth.de <mailto:lyncker@lyth.de>> wrote:
Hi Tihomir,
Thanks for your help , I added the Asteriskparameters as you described below, but I still get the same timeout error: 2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry to 270 seconds. 2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request Timeout [408]. failure #9
Now, my gateway entry looks like the following :
<include> <gateway name="asterisk"> <param name="username" value="28"/> <param name="realm" value="192.168.1.119"/> <param name="proxy" value="192.168.1.119"/> <param name="password" value="test"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="sip-port" value="5060"></param> </gateway> </include>
What can be still wrong here?
Regards,
Filip
Tihomir Culjaga schrieb: > hi Filip, > > > for calling a user... please read this first: > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered... > for making a GW register into e.g. asterisk please use this: > > > <include> > <gateway name="gw01"> > <param name="username" value="USERNAME_ON_ASTERISK"/> > <param name="realm" value="ASTERISK_IP_ADDRESS"/> > <param name="password" value="PASSWORD_ON_ASTERISK"/> > <param name="register" value="true"/> > <param name="caller-id-in-from" value="true"/> > </gateway> > </include> > > this should be enough to register the GW... after that please read > this: > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gate... > > > in your case it will be something like this: > > <extension name="dialGW"> > <condition field="destination_number" > expression="^(NUMBER_TO_SEND_TO_ASTERISK)$"> > <action application="bridge" data="sofia/gateway/gw01/$1"/> > </condition> > </extension> > > > > > > > > > > On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker <lyncker@lyth.de <mailto:lyncker@lyth.de> > <mailto:lyncker@lyth.de <mailto:lyncker@lyth.de>>> wrote: > > Hi List, > > for the first experiments with freeswitch I downloaded the Windows > installation. > Now Im trying to get my 2 Sipphones get connected to. Later I want > connect the freeswitch to my asterisk gateway. > > I find the examples pretty complex therfore Im trying to build up a > simple solution to understand the functions from the scratch .. > > my current problem is , that I cant route my local sips to each > other ( > registration seems to work now). > the next is , that freeshwitch is not able to connect to asterisk. > but I > will describe this later. > > I installed in the Directory a xml file ( called 22.xml) with the > following content : > > <include> > <domain name="$${domain}"> > <user id="22" mailbox="22"> > <params> > <param name="password" value="Xk21%"></param> > <param name="vm-password" value="22"></param> > <param name="sip-port" value="5060"></param> > > </params> > <variables> > <variable name="accountcode" value="22"></variable> > <variable name="user_context" value="default"></variable> > <variable name="effective_caller_id_name" value="Extension > 22"></variable> > <variable name="effective_caller_id_number" > value="22"></variable> > </variables> > </user> > <user id="24" mailbox="24"> > <params> > <param name="password" value="dudeldum"></param> > <param name="vm-password" value="24"></param> > <param name="sip-port" value="5060"></param> > > </params> > <variables> > <variable name="accountcode" value="24"></variable> > <variable name="user_context" value="default"></variable> > <variable name="effective_caller_id_name" value="Extension > 24"></variable> > <variable name="effective_caller_id_number" > value="24"></variable> > </variables> > </user> > </domain> > </include> > > This seems to be ok now. Now I want to dial from 22 to 24 , > wherefore I > configured this dialplan : > > <include> > <context name="any"> > <condition field="destination_number" expression="^(2[0-9])$"> > > <action application="bridge" data="user/${dialed_extension}"/> > > </condition> > </include> > > wich doesnt work , mybe b/c the user/${dialed_extension} I dont > know... > Freeswitch says: > [INFO] switch_core_state_machine.c:136 No Route, Aborting > [NOTICE] switch_core_state_machine.c:137 Hangup > sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34> <mailto:24@192.168.1.34 <mailto:24@192.168.1.34>> > [CS_ROUTING] [NO_ROUTE_DESTINATION] > [NOTICE] switch_core_session.c:1086 Session 17 > (sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34> <mailto:24@192.168.1.34 <mailto:24@192.168.1.34>>) Ended > [NOTICE] switch_core_session.c:1088 Close Channel > sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34> <mailto:24@192.168.1.34 <mailto:24@192.168.1.34>> [CS_DESTROY] > > Im sure , for you guys this cant be a big deal;) > > > Next Point is my Asterisk registration , mybe you can help me out here > to .. : > > In the sip-profiles/external I installed the my_asterisk.xml with that > content : > > <include> > <gateway name="asterisk"> > <param name="username" value="28"></param> > <param name="password" value="test"></param> > <param name="realm" value="28"></param> > <param name="proxy" value="192.168.1.119"></param> > <param name="register" value="true"></param> > </gateway> > </include> > > Freeswitch allways complains a timeout error : > [ERR] sofia_reg.c:1460 asterisk Registration Failed with status > Request > Timeout [408]. failure #17 > [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry > to 540 seconds. > > it seems that It cant connect ( I also tried out to explicit set the > port to 5060 b/c I read something about 5080 .. : <param > name="sip-port" > value="5060"></param> but this didnt help) > In my Asterisk I set in the sip.conf the entry 28 with the pw test > .... > > > If someone could help me with my first steps I would be verrry > thankful ;)) > > cheers > > > Filip > > -- > _________________________________ > Filip Lyncker, Dipl.-Inform. (FH) > > > Lyncker & Theis GmbH > Wilhelmstr. 16 > 65185 Wiesbaden > Germany > > Fon +49 611/9006951 > Fax +49 611/9406125 > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > Steuernummer: 4023897051 > USt-IdNr.: DE255806399 > > Geschäftsführer: > Filip Lyncker, > Armin Theis > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > <mailto:FreeSWITCH-users@lists.freeswitch.org <mailto: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 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto: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 >
-- _________________________________ Filip Lyncker, Dipl.-Inform. (FH)
Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany
Fon +49 611/9006951 Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399
Geschäftsführer: Filip Lyncker, Armin Theis
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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
------------------------------------------------------------------------
_______________________________________________ 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
-- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis
Ok *solved* .... I set in my sip.conf (asterisk) now nat=true, b/c the asterisk ansered the packets sent from lan_ip to the external_ip. now it works, but its not the perfect solution because FS seems to send the packets with an nat envelope or flag. How can i avoid this? the next thing is the dialplan, wich doesnt work at all for me ! ( see my other post with sip registrares) ... if I call now a number , the following entry should route it to my asterisk-gw : <context name="any"> <extension name="dialasterisk"> <condition field="destination_number" expression="^${dialed_extension}$"> <action application="bridge" data="sofia/gateway/asterisk/$1"/> </condition> </extension> </context> but it doesnt and FS says : freeswitch@Bigfish> 2009-09-22 17:10:16.776629 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22@192.168.1.34 [733236b0-be36-0049-8ace-a2903921fd81] 2009-09-22 17:10:16.781511 [INFO] mod_dialplan_xml.c:315 Processing 22->01776721280 in context default 2009-09-22 17:10:16.800065 [NOTICE] switch_ivr.c:1349 Transfer sofia/internal/22@192.168.1.34 to enum[01776721280@default] 2009-09-22 17:10:26.800401 [INFO] switch_core_state_machine.c:136 No Route, Aborting 2009-09-22 17:10:26.800401 [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/22@192.168.1.34 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-09-22 17:10:26.800401 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/internal/22@192.168.1.34) Ended 2009-09-22 17:10:26.800401 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22@192.168.1.34 [CS_DESTROY] what's wrong with my dialplan ? thanks again for help, regards filip Tihomir Culjaga schrieb:
hmmm .. can you register using x-lite or some other softphone with the same credentials?
can you paste a siptrace of the failed registration?
BTW: Make sure nothing is already registered with this credentials when you try with FS
T.
On Tue, Sep 22, 2009 at 12:56 PM, Filip Lyncker <lyncker@lyth.de <mailto:lyncker@lyth.de>> wrote:
Hi Tihomir,
Thanks for your help , I added the Asteriskparameters as you described below, but I still get the same timeout error: 2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry to 270 seconds. 2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request Timeout [408]. failure #9
Now, my gateway entry looks like the following :
<include> <gateway name="asterisk"> <param name="username" value="28"/> <param name="realm" value="192.168.1.119"/> <param name="proxy" value="192.168.1.119"/> <param name="password" value="test"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="sip-port" value="5060"></param> </gateway> </include>
What can be still wrong here?
Regards,
Filip
Tihomir Culjaga schrieb: > hi Filip, > > > for calling a user... please read this first: > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered... > for making a GW register into e.g. asterisk please use this: > > > <include> > <gateway name="gw01"> > <param name="username" value="USERNAME_ON_ASTERISK"/> > <param name="realm" value="ASTERISK_IP_ADDRESS"/> > <param name="password" value="PASSWORD_ON_ASTERISK"/> > <param name="register" value="true"/> > <param name="caller-id-in-from" value="true"/> > </gateway> > </include> > > this should be enough to register the GW... after that please read > this: > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gate... > > > in your case it will be something like this: > > <extension name="dialGW"> > <condition field="destination_number" > expression="^(NUMBER_TO_SEND_TO_ASTERISK)$"> > <action application="bridge" data="sofia/gateway/gw01/$1"/> > </condition> > </extension> > > > > > > > > > > On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker <lyncker@lyth.de <mailto:lyncker@lyth.de> > <mailto:lyncker@lyth.de <mailto:lyncker@lyth.de>>> wrote: > > Hi List, > > for the first experiments with freeswitch I downloaded the Windows > installation. > Now Im trying to get my 2 Sipphones get connected to. Later I want > connect the freeswitch to my asterisk gateway. > > I find the examples pretty complex therfore Im trying to build up a > simple solution to understand the functions from the scratch .. > > my current problem is , that I cant route my local sips to each > other ( > registration seems to work now). > the next is , that freeshwitch is not able to connect to asterisk. > but I > will describe this later. > > I installed in the Directory a xml file ( called 22.xml) with the > following content : > > <include> > <domain name="$${domain}"> > <user id="22" mailbox="22"> > <params> > <param name="password" value="Xk21%"></param> > <param name="vm-password" value="22"></param> > <param name="sip-port" value="5060"></param> > > </params> > <variables> > <variable name="accountcode" value="22"></variable> > <variable name="user_context" value="default"></variable> > <variable name="effective_caller_id_name" value="Extension > 22"></variable> > <variable name="effective_caller_id_number" > value="22"></variable> > </variables> > </user> > <user id="24" mailbox="24"> > <params> > <param name="password" value="dudeldum"></param> > <param name="vm-password" value="24"></param> > <param name="sip-port" value="5060"></param> > > </params> > <variables> > <variable name="accountcode" value="24"></variable> > <variable name="user_context" value="default"></variable> > <variable name="effective_caller_id_name" value="Extension > 24"></variable> > <variable name="effective_caller_id_number" > value="24"></variable> > </variables> > </user> > </domain> > </include> > > This seems to be ok now. Now I want to dial from 22 to 24 , > wherefore I > configured this dialplan : > > <include> > <context name="any"> > <condition field="destination_number" expression="^(2[0-9])$"> > > <action application="bridge" data="user/${dialed_extension}"/> > > </condition> > </include> > > wich doesnt work , mybe b/c the user/${dialed_extension} I dont > know... > Freeswitch says: > [INFO] switch_core_state_machine.c:136 No Route, Aborting > [NOTICE] switch_core_state_machine.c:137 Hangup > sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34> <mailto:24@192.168.1.34 <mailto:24@192.168.1.34>> > [CS_ROUTING] [NO_ROUTE_DESTINATION] > [NOTICE] switch_core_session.c:1086 Session 17 > (sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34> <mailto:24@192.168.1.34 <mailto:24@192.168.1.34>>) Ended > [NOTICE] switch_core_session.c:1088 Close Channel > sofia/internal/24@192.168.1.34 <mailto:24@192.168.1.34> <mailto:24@192.168.1.34 <mailto:24@192.168.1.34>> [CS_DESTROY] > > Im sure , for you guys this cant be a big deal;) > > > Next Point is my Asterisk registration , mybe you can help me out here > to .. : > > In the sip-profiles/external I installed the my_asterisk.xml with that > content : > > <include> > <gateway name="asterisk"> > <param name="username" value="28"></param> > <param name="password" value="test"></param> > <param name="realm" value="28"></param> > <param name="proxy" value="192.168.1.119"></param> > <param name="register" value="true"></param> > </gateway> > </include> > > Freeswitch allways complains a timeout error : > [ERR] sofia_reg.c:1460 asterisk Registration Failed with status > Request > Timeout [408]. failure #17 > [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry > to 540 seconds. > > it seems that It cant connect ( I also tried out to explicit set the > port to 5060 b/c I read something about 5080 .. : <param > name="sip-port" > value="5060"></param> but this didnt help) > In my Asterisk I set in the sip.conf the entry 28 with the pw test > .... > > > If someone could help me with my first steps I would be verrry > thankful ;)) > > cheers > > > Filip > > -- > _________________________________ > Filip Lyncker, Dipl.-Inform. (FH) > > > Lyncker & Theis GmbH > Wilhelmstr. 16 > 65185 Wiesbaden > Germany > > Fon +49 611/9006951 > Fax +49 611/9406125 > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > Steuernummer: 4023897051 > USt-IdNr.: DE255806399 > > Geschäftsführer: > Filip Lyncker, > Armin Theis > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > <mailto:FreeSWITCH-users@lists.freeswitch.org <mailto: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 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto: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 >
-- _________________________________ Filip Lyncker, Dipl.-Inform. (FH)
Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany
Fon +49 611/9006951 Fax +49 611/9406125
Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399
Geschäftsführer: Filip Lyncker, Armin Theis
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto: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
------------------------------------------------------------------------
_______________________________________________ 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
-- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis
participants (2)
-
Filip Lyncker
-
Tihomir Culjaga