Call send from Freeswitch to Asterisk
Hello All, I am using Freeswitch on Debain 8. Configured few DIDs. Calls coming from provider and I am trying to send Call to a Asterisk server but getting an error "MANDATORY_IE_MISSING". DID Provider (3.3.3.3) => FS (192.168.1.70) => Asterisk (192.168.1.92) => Ext 8001 *FS dialplan =>* <extension name="public"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/external/$1@192.168.1.92"/> <action application="hangup"/> </condition> </extension> Added DID Provider IP and Asterisk IP in ACL. When I checked on Asterisk, got an error in sngrep, "401 Unauthorized". Attached image is of error. *On Asterisk =>* SIP.conf [freeswitch_1] type=peer host=192.168.1.70 port=5080 disallow=all allow=ulaw trustrpid=yes sendrpid=yes context=from-internal EXTENSIONS.conf exten => _XXX.,1,Dial(SIP/8001,30,oT) exten => _XXX.,n,Hangup Tried on port 5080 and 5060. Error both time same. My point is, this is Asterisk issue or Freeswitch? Please advice. -- Thanks & Regard Uday. Mobile:- +91-9377579349
Hi, I can see type=peer, i guess that is only to put calls on the IP, you need to either use type friend or user. Regards Abbasi On Tue, Oct 31, 2017 at 4:46 PM, Uday kumar <udy786@gmail.com> wrote:
Hello All,
I am using Freeswitch on Debain 8. Configured few DIDs. Calls coming from provider and I am trying to send Call to a Asterisk server but getting an error "MANDATORY_IE_MISSING".
DID Provider (3.3.3.3) => FS (192.168.1.70) => Asterisk (192.168.1.92) => Ext 8001
*FS dialplan =>*
<extension name="public"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/external/$1@192.168.1.92"/> <action application="hangup"/> </condition> </extension>
Added DID Provider IP and Asterisk IP in ACL.
When I checked on Asterisk, got an error in sngrep, "401 Unauthorized". Attached image is of error.
*On Asterisk =>*
SIP.conf
[freeswitch_1] type=peer host=192.168.1.70 port=5080 disallow=all allow=ulaw trustrpid=yes sendrpid=yes context=from-internal
EXTENSIONS.conf
exten => _XXX.,1,Dial(SIP/8001,30,oT) exten => _XXX.,n,Hangup
Tried on port 5080 and 5060. Error both time same.
My point is, this is Asterisk issue or Freeswitch? Please advice.
-- Thanks & Regard Uday. Mobile:- +91-9377579349
_________________________________________________________________________ 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
Hello Abbasi, Thank you for your reply. I tried in both cases, type=friend and type=user but still same error. Thanks Uday. Skype:- uday.kumar786 On Tue, Oct 31, 2017 at 10:37 PM, Bilal Abbasi <bilaln018@gmail.com> wrote:
Hi, I can see type=peer, i guess that is only to put calls on the IP, you need to either use type friend or user.
Regards Abbasi
On Tue, Oct 31, 2017 at 4:46 PM, Uday kumar <udy786@gmail.com> wrote:
Hello All,
I am using Freeswitch on Debain 8. Configured few DIDs. Calls coming from provider and I am trying to send Call to a Asterisk server but getting an error "MANDATORY_IE_MISSING".
DID Provider (3.3.3.3) => FS (192.168.1.70) => Asterisk (192.168.1.92) => Ext 8001
*FS dialplan =>*
<extension name="public"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/external/$1@192.168.1.92"/> <action application="hangup"/> </condition> </extension>
Added DID Provider IP and Asterisk IP in ACL.
When I checked on Asterisk, got an error in sngrep, "401 Unauthorized". Attached image is of error.
*On Asterisk =>*
SIP.conf
[freeswitch_1] type=peer host=192.168.1.70 port=5080 disallow=all allow=ulaw trustrpid=yes sendrpid=yes context=from-internal
EXTENSIONS.conf
exten => _XXX.,1,Dial(SIP/8001,30,oT) exten => _XXX.,n,Hangup
Tried on port 5080 and 5060. Error both time same.
My point is, this is Asterisk issue or Freeswitch? Please advice.
-- Thanks & Regard Uday. Mobile:- +91-9377579349 <093775%2079349>
_________________________________________________________________________ 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 & Regard Uday. Mobile:- +91-9377579349
You probably need insecure directive in your FS peer in asterisk https://www.voip-info.org/wiki/view/Asterisk+sip+insecure On Tue, Oct 31, 2017 at 11:46 AM, Uday kumar <udy786@gmail.com> wrote:
Hello All,
I am using Freeswitch on Debain 8. Configured few DIDs. Calls coming from provider and I am trying to send Call to a Asterisk server but getting an error "MANDATORY_IE_MISSING".
DID Provider (3.3.3.3) => FS (192.168.1.70) => Asterisk (192.168.1.92) => Ext 8001
FS dialplan =>
<extension name="public"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/external/$1@192.168.1.92"/> <action application="hangup"/> </condition> </extension>
Added DID Provider IP and Asterisk IP in ACL.
When I checked on Asterisk, got an error in sngrep, "401 Unauthorized". Attached image is of error.
On Asterisk =>
SIP.conf
[freeswitch_1] type=peer host=192.168.1.70 port=5080 disallow=all allow=ulaw trustrpid=yes sendrpid=yes context=from-internal
EXTENSIONS.conf
exten => _XXX.,1,Dial(SIP/8001,30,oT) exten => _XXX.,n,Hangup
Tried on port 5080 and 5060. Error both time same.
My point is, this is Asterisk issue or Freeswitch? Please advice.
-- Thanks & Regard Uday. Mobile:- +91-9377579349
_________________________________________________________________________ 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 (3)
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Bilal Abbasi
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Brian :
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Uday kumar