Hi, We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server. Call flow is as below: User => opensips => FS => PBX, but call not sending out from FS and drop by FS. Here I am attaching both sip logs and freeswitch logs. Can anyone please help to sort out the issue. -- Thanks, Rutu Patel <http://www.inextrix.com>
Hi Rutu, In logs I can see that user-agent Asterisk is sending INVITE to Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This a bit differs from what you explained in your mail... First think what pop up in my mind is that you need to configure Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my first guess... https://freeswitch.org/confluence/display/FREESWITCH/Asterisk With kind regards, Jurijs On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server.
Call flow is as below:
User => opensips => FS => PBX,
but call not sending out from FS and drop by FS.
Here I am attaching both sip logs and freeswitch logs.
Can anyone please help to sort out the issue. -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
You need to supply auth credentials to Asterisk because its set to authenticate calls. The best thing to do would be to make a gateway pointing to Asterisk with all the credentials and call it using that. On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi Rutu,
In logs I can see that user-agent Asterisk is sending INVITE to Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This a bit differs from what you explained in your mail...
First think what pop up in my mind is that you need to configure Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my first guess...
https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
With kind regards,
Jurijs
On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server.
Call flow is as below:
User => opensips => FS => PBX,
but call not sending out from FS and drop by FS.
Here I am attaching both sip logs and freeswitch logs.
Can anyone please help to sort out the issue. -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
Hi, Thank you for reply. We have registered extension of Freeswitch in Asterisk as below register=>test-ext:secret@192.168.1.64/test-ext We want to send call from Freeswitch to Asterisk. What authentication parameters we need to set in Freeswitch ? -- Thanks, Rutu Patel On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You need to supply auth credentials to Asterisk because its set to authenticate calls. The best thing to do would be to make a gateway pointing to Asterisk with all the credentials and call it using that.
On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi Rutu,
In logs I can see that user-agent Asterisk is sending INVITE to Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This a bit differs from what you explained in your mail...
First think what pop up in my mind is that you need to configure Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my first guess...
https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
With kind regards,
Jurijs
On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server.
Call flow is as below:
User => opensips => FS => PBX,
but call not sending out from FS and drop by FS.
Here I am attaching both sip logs and freeswitch logs.
Can anyone please help to sort out the issue. -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
Try a gateway. Look in conf/sip_profiles/external/example.xml for a heavily-commented sample. For more detailed discussion see this nice doc: https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration -MSC On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
Thank you for reply.
We have registered extension of Freeswitch in Asterisk as below
register=>test-ext:secret@192.168.1.64/test-ext
We want to send call from Freeswitch to Asterisk.
What authentication parameters we need to set in Freeswitch ?
-- Thanks, Rutu Patel On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You need to supply auth credentials to Asterisk because its set to authenticate calls. The best thing to do would be to make a gateway pointing to Asterisk with all the credentials and call it using that.
On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi Rutu,
In logs I can see that user-agent Asterisk is sending INVITE to Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This a bit differs from what you explained in your mail...
First think what pop up in my mind is that you need to configure Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my first guess...
https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
With kind regards,
Jurijs
On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server.
Call flow is as below:
User => opensips => FS => PBX,
but call not sending out from FS and drop by FS.
Here I am attaching both sip logs and freeswitch logs.
Can anyone please help to sort out the issue. -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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 Michael Thank you for your assistance. We are trying to make call between two freeswitch users. One user is registered in softphone and other is registered in Asterisk by register string but asterisk user cannot authenticate properly and freeswitch directly send 480 Temporary Unavailable. In sip profile we set challenge-realm= auto_to also tried with auto_from but still not getting success. I am attaching some more logs if anyone can guide in right direction. -- Thanks, Rutu Patel On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc@freeswitch.org> wrote:
Try a gateway. Look in conf/sip_profiles/external/example.xml for a heavily-commented sample. For more detailed discussion see this nice doc: https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration
-MSC
On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
Thank you for reply.
We have registered extension of Freeswitch in Asterisk as below
register=>test-ext:secret@192.168.1.64/test-ext
We want to send call from Freeswitch to Asterisk.
What authentication parameters we need to set in Freeswitch ?
-- Thanks, Rutu Patel On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You need to supply auth credentials to Asterisk because its set to authenticate calls. The best thing to do would be to make a gateway pointing to Asterisk with all the credentials and call it using that.
On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga <jurijs.ivolga@gmail.com
wrote:
Hi Rutu,
In logs I can see that user-agent Asterisk is sending INVITE to Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This a bit differs from what you explained in your mail...
First think what pop up in my mind is that you need to configure Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my first guess...
https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
With kind regards,
Jurijs
On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server.
Call flow is as below:
User => opensips => FS => PBX,
but call not sending out from FS and drop by FS.
Here I am attaching both sip logs and freeswitch logs.
Can anyone please help to sort out the issue. -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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 Rutu, Just a shot in a dark. :) Did you added sip peer record in sip.conf, something similar to: [freeswitch] type=peer host=IP_ADDRESS_OF_FREESWITCH_SERVER username=HOSTNAME.DOMAIN.COM port=5080 fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER secret=BOOTH_WAY_PASSWORD With kind regards, Jurijs On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi Michael
Thank you for your assistance.
We are trying to make call between two freeswitch users.
One user is registered in softphone and other is registered in Asterisk by register string but asterisk user cannot authenticate properly and freeswitch directly send 480 Temporary Unavailable.
In sip profile we set challenge-realm= auto_to also tried with auto_from but still not getting success.
I am attaching some more logs if anyone can guide in right direction.
-- Thanks, Rutu Patel
On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc@freeswitch.org> wrote:
Try a gateway. Look in conf/sip_profiles/external/example.xml for a heavily-commented sample. For more detailed discussion see this nice doc:
https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration
-MSC
On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
Thank you for reply.
We have registered extension of Freeswitch in Asterisk as below
register=>test-ext:secret@192.168.1.64/test-ext
We want to send call from Freeswitch to Asterisk.
What authentication parameters we need to set in Freeswitch ?
-- Thanks, Rutu Patel On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You need to supply auth credentials to Asterisk because its set to authenticate calls. The best thing to do would be to make a gateway pointing to Asterisk with all the credentials and call it using that.
On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga < jurijs.ivolga@gmail.com> wrote:
Hi Rutu,
In logs I can see that user-agent Asterisk is sending INVITE to Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This a bit differs from what you explained in your mail...
First think what pop up in my mind is that you need to configure Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my first guess...
https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
With kind regards,
Jurijs
On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server.
Call flow is as below:
User => opensips => FS => PBX,
but call not sending out from FS and drop by FS.
Here I am attaching both sip logs and freeswitch logs.
Can anyone please help to sort out the issue. -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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 Juris, Yes we added peer in asterisk sip.conf. Are we still missing any configurations? On Thursday, March 31, 2016, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi Rutu,
Just a shot in a dark. :)
Did you added sip peer record in sip.conf, something similar to:
[freeswitch] type=peer host=IP_ADDRESS_OF_FREESWITCH_SERVER username=HOSTNAME.DOMAIN.COM port=5080 fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER secret=BOOTH_WAY_PASSWORD
With kind regards,
Jurijs
On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <rutu.patel@inextrix.com <javascript:_e(%7B%7D,'cvml','rutu.patel@inextrix.com');>> wrote:
Hi Michael
Thank you for your assistance.
We are trying to make call between two freeswitch users.
One user is registered in softphone and other is registered in Asterisk by register string but asterisk user cannot authenticate properly and freeswitch directly send 480 Temporary Unavailable.
In sip profile we set challenge-realm= auto_to also tried with auto_from but still not getting success.
I am attaching some more logs if anyone can guide in right direction.
-- Thanks, Rutu Patel
On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc@freeswitch.org <javascript:_e(%7B%7D,'cvml','msc@freeswitch.org');>> wrote:
Try a gateway. Look in conf/sip_profiles/external/example.xml for a heavily-commented sample. For more detailed discussion see this nice doc:
https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration
-MSC
On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel@inextrix.com <javascript:_e(%7B%7D,'cvml','rutu.patel@inextrix.com');>> wrote:
Hi,
Thank you for reply.
We have registered extension of Freeswitch in Asterisk as below
register=>test-ext:secret@192.168.1.64/test-ext
We want to send call from Freeswitch to Asterisk.
What authentication parameters we need to set in Freeswitch ?
-- Thanks, Rutu Patel On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale < anthony.minessale@gmail.com <javascript:_e(%7B%7D,'cvml','anthony.minessale@gmail.com');>> wrote:
You need to supply auth credentials to Asterisk because its set to authenticate calls. The best thing to do would be to make a gateway pointing to Asterisk with all the credentials and call it using that.
On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga < jurijs.ivolga@gmail.com <javascript:_e(%7B%7D,'cvml','jurijs.ivolga@gmail.com');>> wrote:
Hi Rutu,
In logs I can see that user-agent Asterisk is sending INVITE to Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This a bit differs from what you explained in your mail...
First think what pop up in my mind is that you need to configure Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my first guess...
https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
With kind regards,
Jurijs
On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel@inextrix.com <javascript:_e(%7B%7D,'cvml','rutu.patel@inextrix.com');>> wrote:
> Hi, > > We are sending call to asterisk PBX but we are getting 480 > Temporarily Unavailable immediately. from freeswitch server. > > Call flow is as below: > > User => opensips => FS => PBX, > > but call not sending out from FS and drop by FS. > > Here I am attaching both sip logs and freeswitch logs. > > Can anyone please help to sort out the issue. > -- > Thanks, > Rutu Patel > <http://www.inextrix.com> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > <javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > <javascript:_e(%7B%7D,'cvml','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 >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','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
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org <javascript:_e(%7B%7D,'cvml','sip:888@conference.freeswitch.org');> ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','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
-- -- Thanks, Rutu Patel iNextrix Technologies Pvt. Ltd. www.inextrix.com
Hi, I'm not sure about your environment, but on this point I will recommend you to isolate issue. So on this point you should understand if your sip trunk between Asterisk and Freeswitch configured correctly, without involving Opensips and etc. To test SIP trunk you can try to send call directly from freeswitch to Asterisk and see if it works, you can send call just directly from freeswitch console, something similar to: originate sofia/example/300@foo.com 8600 https://freeswitch.org/confluence/display/FREESWITCH/mod_commands If you call will reach Asterisk without any issue, then this means that sip trunk created correctly and issue is somewhere on Opensips side or any other part of your set-up. Second option will be to try to configure sip trunk without password and check if it helps https://freeswitch.org/confluence/display/FREESWITCH/Asterisk With kind regards, Jurijs On Fri, Apr 1, 2016 at 12:54 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi Juris,
Yes we added peer in asterisk sip.conf.
Are we still missing any configurations?
On Thursday, March 31, 2016, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi Rutu,
Just a shot in a dark. :)
Did you added sip peer record in sip.conf, something similar to:
[freeswitch] type=peer host=IP_ADDRESS_OF_FREESWITCH_SERVER username=HOSTNAME.DOMAIN.COM port=5080 fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER secret=BOOTH_WAY_PASSWORD
With kind regards,
Jurijs
On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi Michael
Thank you for your assistance.
We are trying to make call between two freeswitch users.
One user is registered in softphone and other is registered in Asterisk by register string but asterisk user cannot authenticate properly and freeswitch directly send 480 Temporary Unavailable.
In sip profile we set challenge-realm= auto_to also tried with auto_from but still not getting success.
I am attaching some more logs if anyone can guide in right direction.
-- Thanks, Rutu Patel
On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc@freeswitch.org> wrote:
Try a gateway. Look in conf/sip_profiles/external/example.xml for a heavily-commented sample. For more detailed discussion see this nice doc:
https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration
-MSC
On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
Thank you for reply.
We have registered extension of Freeswitch in Asterisk as below
register=>test-ext:secret@192.168.1.64/test-ext
We want to send call from Freeswitch to Asterisk.
What authentication parameters we need to set in Freeswitch ?
-- Thanks, Rutu Patel On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You need to supply auth credentials to Asterisk because its set to authenticate calls. The best thing to do would be to make a gateway pointing to Asterisk with all the credentials and call it using that.
On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga < jurijs.ivolga@gmail.com> wrote:
> Hi Rutu, > > In logs I can see that user-agent Asterisk is sending INVITE to > Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This > a bit differs from what you explained in your mail... > > First think what pop up in my mind is that you need to configure > Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my > first guess... > > https://freeswitch.org/confluence/display/FREESWITCH/Asterisk > > With kind regards, > > Jurijs > > On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel@inextrix.com > > wrote: > >> Hi, >> >> We are sending call to asterisk PBX but we are getting 480 >> Temporarily Unavailable immediately. from freeswitch server. >> >> Call flow is as below: >> >> User => opensips => FS => PBX, >> >> but call not sending out from FS and drop by FS. >> >> Here I am attaching both sip logs and freeswitch logs. >> >> Can anyone please help to sort out the issue. >> -- >> Thanks, >> Rutu Patel >> <http://www.inextrix.com> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > 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 >
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- -- Thanks, Rutu Patel iNextrix Technologies Pvt. Ltd. www.inextrix.com
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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, Thank you for your help to debug the issue. We are now trying without opensips but still facing authentication issue. We created two users of freeswitch in asterisk as peer. For example, First peer in sip.conf is 456789 and second is 123456. [123456] type=friend host=x.x.x.45 username=123456 secret=abcdef disallow=all allow=g729 trustrpid=yes canreinvite=yes sendrpid=yes context=extensions-test qualify=yes directmedia=yes Now when we are trying to reach 123456, the request reach to asterisk but not authenticate proper user(123456). It authenticate first peer(456789) defined in sip.conf. If we define 123456 before 456789 then it works but we have multiple peers in asterisk so how can we manage that ? We also tried by sending user credential direct in bridge,but still it is authenticate the first peer defined in sip.conf . {sip_auth_username=123456,sip_auth_password=abcdef,sip_contact_user=123456}sofia/default/NUMBER@IP Can anyone please explain how to authenticate specific peer ? Please let me know if you want any other details. -- Thanks, Rutu Patel <http://www.inextrix.com> On Fri, Apr 1, 2016 at 3:56 PM, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi,
I'm not sure about your environment, but on this point I will recommend you to isolate issue. So on this point you should understand if your sip trunk between Asterisk and Freeswitch configured correctly, without involving Opensips and etc. To test SIP trunk you can try to send call directly from freeswitch to Asterisk and see if it works, you can send call just directly from freeswitch console, something similar to:
originate sofia/example/300@foo.com 8600
https://freeswitch.org/confluence/display/FREESWITCH/mod_commands
If you call will reach Asterisk without any issue, then this means that sip trunk created correctly and issue is somewhere on Opensips side or any other part of your set-up.
Second option will be to try to configure sip trunk without password and check if it helps
https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
With kind regards,
Jurijs
On Fri, Apr 1, 2016 at 12:54 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi Juris,
Yes we added peer in asterisk sip.conf.
Are we still missing any configurations?
On Thursday, March 31, 2016, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi Rutu,
Just a shot in a dark. :)
Did you added sip peer record in sip.conf, something similar to:
[freeswitch] type=peer host=IP_ADDRESS_OF_FREESWITCH_SERVER username=HOSTNAME.DOMAIN.COM port=5080 fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER secret=BOOTH_WAY_PASSWORD
With kind regards,
Jurijs
On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi Michael
Thank you for your assistance.
We are trying to make call between two freeswitch users.
One user is registered in softphone and other is registered in Asterisk by register string but asterisk user cannot authenticate properly and freeswitch directly send 480 Temporary Unavailable.
In sip profile we set challenge-realm= auto_to also tried with auto_from but still not getting success.
I am attaching some more logs if anyone can guide in right direction.
-- Thanks, Rutu Patel
On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc@freeswitch.org> wrote:
Try a gateway. Look in conf/sip_profiles/external/example.xml for a heavily-commented sample. For more detailed discussion see this nice doc:
https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration
-MSC
On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
Thank you for reply.
We have registered extension of Freeswitch in Asterisk as below
register=>test-ext:secret@192.168.1.64/test-ext
We want to send call from Freeswitch to Asterisk.
What authentication parameters we need to set in Freeswitch ?
-- Thanks, Rutu Patel On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
> You need to supply auth credentials to Asterisk because its set to > authenticate calls. > The best thing to do would be to make a gateway pointing to Asterisk > with all the credentials and call it using that. > > > > > On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga < > jurijs.ivolga@gmail.com> wrote: > >> Hi Rutu, >> >> In logs I can see that user-agent Asterisk is sending INVITE to >> Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This >> a bit differs from what you explained in your mail... >> >> First think what pop up in my mind is that you need to configure >> Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my >> first guess... >> >> https://freeswitch.org/confluence/display/FREESWITCH/Asterisk >> >> With kind regards, >> >> Jurijs >> >> On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel < >> rutu.patel@inextrix.com> wrote: >> >>> Hi, >>> >>> We are sending call to asterisk PBX but we are getting 480 >>> Temporarily Unavailable immediately. from freeswitch server. >>> >>> Call flow is as below: >>> >>> User => opensips => FS => PBX, >>> >>> but call not sending out from FS and drop by FS. >>> >>> Here I am attaching both sip logs and freeswitch logs. >>> >>> Can anyone please help to sort out the issue. >>> -- >>> Thanks, >>> Rutu Patel >>> <http://www.inextrix.com> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > 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 >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- -- Thanks, Rutu Patel iNextrix Technologies Pvt. Ltd. www.inextrix.com
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
Can anyone please help ? -- Thanks, Rutu Patel <http://www.inextrix.com> On Tue, Apr 5, 2016 at 10:46 AM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi,
Thank you for your help to debug the issue.
We are now trying without opensips but still facing authentication issue.
We created two users of freeswitch in asterisk as peer.
For example, First peer in sip.conf is 456789 and second is 123456.
[123456] type=friend host=x.x.x.45 username=123456 secret=abcdef disallow=all allow=g729 trustrpid=yes canreinvite=yes sendrpid=yes context=extensions-test qualify=yes directmedia=yes
Now when we are trying to reach 123456, the request reach to asterisk but not authenticate proper user(123456). It authenticate first peer(456789) defined in sip.conf.
If we define 123456 before 456789 then it works but we have multiple peers in asterisk so how can we manage that ?
We also tried by sending user credential direct in bridge,but still it is authenticate the first peer defined in sip.conf .
{sip_auth_username=123456,sip_auth_password=abcdef,sip_contact_user=123456}sofia/default/NUMBER@IP
Can anyone please explain how to authenticate specific peer ?
Please let me know if you want any other details.
-- Thanks, Rutu Patel
On Fri, Apr 1, 2016 at 3:56 PM, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi,
I'm not sure about your environment, but on this point I will recommend you to isolate issue. So on this point you should understand if your sip trunk between Asterisk and Freeswitch configured correctly, without involving Opensips and etc. To test SIP trunk you can try to send call directly from freeswitch to Asterisk and see if it works, you can send call just directly from freeswitch console, something similar to:
originate sofia/example/300@foo.com 8600
https://freeswitch.org/confluence/display/FREESWITCH/mod_commands
If you call will reach Asterisk without any issue, then this means that sip trunk created correctly and issue is somewhere on Opensips side or any other part of your set-up.
Second option will be to try to configure sip trunk without password and check if it helps
https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
With kind regards,
Jurijs
On Fri, Apr 1, 2016 at 12:54 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi Juris,
Yes we added peer in asterisk sip.conf.
Are we still missing any configurations?
On Thursday, March 31, 2016, Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:
Hi Rutu,
Just a shot in a dark. :)
Did you added sip peer record in sip.conf, something similar to:
[freeswitch] type=peer host=IP_ADDRESS_OF_FREESWITCH_SERVER username=HOSTNAME.DOMAIN.COM port=5080 fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER secret=BOOTH_WAY_PASSWORD
With kind regards,
Jurijs
On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hi Michael
Thank you for your assistance.
We are trying to make call between two freeswitch users.
One user is registered in softphone and other is registered in Asterisk by register string but asterisk user cannot authenticate properly and freeswitch directly send 480 Temporary Unavailable.
In sip profile we set challenge-realm= auto_to also tried with auto_from but still not getting success.
I am attaching some more logs if anyone can guide in right direction.
-- Thanks, Rutu Patel
On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc@freeswitch.org> wrote:
Try a gateway. Look in conf/sip_profiles/external/example.xml for a heavily-commented sample. For more detailed discussion see this nice doc:
https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration
-MSC
On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel@inextrix.com > wrote:
> Hi, > > Thank you for reply. > > We have registered extension of Freeswitch in Asterisk as below > > register=>test-ext:secret@192.168.1.64/test-ext > > We want to send call from Freeswitch to Asterisk. > > What authentication parameters we need to set in Freeswitch ? > > -- > Thanks, > Rutu Patel > On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale < > anthony.minessale@gmail.com> wrote: > >> You need to supply auth credentials to Asterisk because its set to >> authenticate calls. >> The best thing to do would be to make a gateway pointing to >> Asterisk with all the credentials and call it using that. >> >> >> >> >> On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga < >> jurijs.ivolga@gmail.com> wrote: >> >>> Hi Rutu, >>> >>> In logs I can see that user-agent Asterisk is sending INVITE to >>> Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This >>> a bit differs from what you explained in your mail... >>> >>> First think what pop up in my mind is that you need to configure >>> Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my >>> first guess... >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/Asterisk >>> >>> With kind regards, >>> >>> Jurijs >>> >>> On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel < >>> rutu.patel@inextrix.com> wrote: >>> >>>> Hi, >>>> >>>> We are sending call to asterisk PBX but we are getting 480 >>>> Temporarily Unavailable immediately. from freeswitch server. >>>> >>>> Call flow is as below: >>>> >>>> User => opensips => FS => PBX, >>>> >>>> but call not sending out from FS and drop by FS. >>>> >>>> Here I am attaching both sip logs and freeswitch logs. >>>> >>>> Can anyone please help to sort out the issue. >>>> -- >>>> Thanks, >>>> Rutu Patel >>>> <http://www.inextrix.com> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> >> >> -- >> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> <http://freeswitch.org/g+>* >> >> ClueCon Weekly Development Call >> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > 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 >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- -- Thanks, Rutu Patel iNextrix Technologies Pvt. Ltd. www.inextrix.com
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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 (4)
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Anthony Minessale
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Jurijs Ivolga
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Michael Collins
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Rutu Patel