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lyncker at lyth.de
Guest
|
 Posted: Tue Sep 22, 2009 6:08 am Post subject: [Freeswitch-users] Some Newbie questions about dialplan and |
 |
|
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:
| Quote: | hi Filip,
for calling a user... please read this first:
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User
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_gateways
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
_______________________________________________
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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 |
lyncker at lyth.de
Guest
|
| Posted: Tue Sep 22, 2009 10:55 am Post subject: [Freeswitch-users] Some Newbie questions about dialplan and |
|
|
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:
| Quote: | 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:
| Quote: | hi Filip,
for calling a user... please read this first:
|
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User
| Quote: | 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_gateways
| Quote: |
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>
| Quote: | <mailto:lyncker@lyth.de <mailto:lyncker@lyth.de>>> wrote:
Hi List,
for the first experiments with freeswitch I downloaded the
|
Windows
| Quote: | installation.
Now Im trying to get my 2 Sipphones get connected to. Later
|
I want
| Quote: | connect the freeswitch to my asterisk gateway.
I find the examples pretty complex therfore Im trying to
|
build up a
| Quote: | 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.
| Quote: | but I
will describe this later.
I installed in the Directory a xml file ( called 22.xml)
|
with the
| Quote: | 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}"/>
| Quote: |
</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>>
| Quote: | [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
| Quote: | [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]
| Quote: |
Im sure , for you guys this cant be a big deal;)
Next Point is my Asterisk registration , mybe you can help
|
me out here
| Quote: | to .. :
In the sip-profiles/external I installed the my_asterisk.xml
|
with that
| Quote: | 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
| Quote: | to 540 seconds.
it seems that It cant connect ( I also tried out to explicit
|
set the
| Quote: | 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
| Quote: | ....
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>
| Quote: | <mailto:FreeSWITCH-users@lists.freeswitch.org
|
<mailto:FreeSWITCH-users@lists.freeswitch.org>>
| Quote: | http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
|
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
| Quote: | http://www.freeswitch.org
|
------------------------------------------------------------------------
<mailto:FreeSWITCH-users@lists.freeswitch.org>
| Quote: | http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
|
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
| Quote: | 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
_______________________________________________
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 |
|
| Back to top |
|
|
lyncker at lyth.de
Guest
|
| Posted: Tue Sep 22, 2009 10:58 am Post subject: [Freeswitch-users] Some Newbie questions about dialplan and |
|
|
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:
| Quote: | 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:
| Quote: | hi Filip,
for calling a user... please read this first:
|
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User
| Quote: | 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_gateways
| Quote: |
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>
| Quote: | <mailto:lyncker@lyth.de <mailto:lyncker@lyth.de>>> wrote:
Hi List,
for the first experiments with freeswitch I downloaded the
|
Windows
| Quote: | installation.
Now Im trying to get my 2 Sipphones get connected to. Later
|
I want
| Quote: | connect the freeswitch to my asterisk gateway.
I find the examples pretty complex therfore Im trying to
|
build up a
| Quote: | 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.
| Quote: | but I
will describe this later.
I installed in the Directory a xml file ( called 22.xml)
|
with the
| Quote: | 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}"/>
| Quote: |
</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>>
| Quote: | [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
| Quote: | [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]
| Quote: |
Im sure , for you guys this cant be a big deal;)
Next Point is my Asterisk registration , mybe you can help
|
me out here
| Quote: | to .. :
In the sip-profiles/external I installed the my_asterisk.xml
|
with that
| Quote: | 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
| Quote: | to 540 seconds.
it seems that It cant connect ( I also tried out to explicit
|
set the
| Quote: | 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
| Quote: | ....
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
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FreeSWITCH-users@lists.freeswitch.org
|
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| Quote: | <mailto:FreeSWITCH-users@lists.freeswitch.org
|
<mailto:FreeSWITCH-users@lists.freeswitch.org>>
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--
_________________________________
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
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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
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