Configure Asterisk to dial through 3cx

[jessydm]

Hi

Thanks

 

[leejor]

I do not want to create SIP trunk on my asterisk server at this time.

But you had to create a SIP trunk to appear as an extension to 3CX.
This is registering properly as an extension on the 3Cx end? Does a SIP device (ATA or softphone) set up as the same extension as the incoming Asterisk extension work OK?

 

[jessydm]

I do not want to create SIP trunk on my asterisk server at this time.

But you had to create a SIP trunk to appear as an extension to 3CX.
This is registering properly as an extension on the 3Cx end? Does a SIP device (ATA or softphone) set up as the same extension as the incoming Asterisk extension work OK?

Sorry for not clearing properly. I do have created sip trunk on asterisk that registers a extension on 3cx. It appears online and registered in extension status windows. The problem is not being able to dial out...
The SIP trunk on 3cx is working fine and i can dial in and out from a 3cx extension. but not from asterisk extension. As you could see from log , it is having authentication issues at 3cx end.

 

[jessydm]

looks to me problem might be somewhere here

23:16:06.406 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification

 

[SY]

looks to me problem might be somewhere here

23:16:06.406 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification

You can compare invite request provided by "Asterisk extension" and invite requests from other extensions which are working properly.
I'm sure you will get the answer. PBX doesn't differentiate extensions.

 

[jessydm]

How can i see sip registration information...

looks to me like 3cx is blocking asterisk to call through..

I can dialout with ATA and soft phones when registered directly to 3cx as extension but not through asterisk box...
here's the server log

22:43:41.140  [CM503013]: Call(33): Incoming call rejected, caller is unknown; msg=SipReq:  INVITE [email protected] tid=28a2e450 cseq=INVITE [email protected] / 103 from(wire)
22:43:41.031  [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
22:43:41.031  [CM302001]: Authorization system can not identify source of: SipReq:  INVITE [email protected] tid=00730770 cseq=INVITE [email protected] / 102 from(wire)

where can i find more information about these above error...what is causing the error ??

 

[N_G]

We dont block any calls from Asterisk. We are open to interoperability and we have installs that using them in conjunction. The way to do this is to setup a SIP trunk in Asterisk to 3CX.

You might require product support. If you have a commercial edition and support package, then you can open a support ticket for help. Note that for the Asterisk side you have to open a ticket with Asterisk. But we are ready to help on the 3CX side if you setup a SIP trunk.

 

[galega]

We have an asterisk machine working as a "gateway" for 3cx in a testing environment and It's working fine!

 

[jessydm]

We have an asterisk machine working as a "gateway" for 3cx in a testing environment and It's working fine!

would you please ellobarate your asterisk sip.conf section regarding to setting up trunk to 3cx.

 

[SY]

How can i see sip registration information...

looks to me like 3cx is blocking asterisk to call through..

I can dialout with ATA and soft phones when registered directly to 3cx as extension but not through asterisk box...
here's the server log

22:43:41.140  [CM503013]: Call(33): Incoming call rejected, caller is unknown; msg=SipReq:  INVITE [email protected] tid=28a2e450 cseq=INVITE [email protected] / 103 from(wire)
22:43:41.031  [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
22:43:41.031  [CM302001]: Authorization system can not identify source of: SipReq:  INVITE [email protected] tid=00730770 cseq=INVITE [email protected] / 102 from(wire)

where can i find more information about these above error...what is causing the error ??

Hint: compare "from" header provided by the "phones registered directly to 3cx" with "from" header provided by your extension. Use wireshark. It will help us to avoid miscommunication.

Thanks
P.S. We don't block anything. Could you please try to leash your fantasy?

 

[abundisnet]

I don´t know much about asterisk, but after read youyr post last night I started a test in elastix with 3CX.
Now I can call any extension on 3CX PBX, but at this moment I can´t (even) call a lines outside the 3CX PBX.

I hope soon can resolve this problem.

 

[abundisnet]

After try&mistake, I found the following:
I give the Elastix PBX the extension 103 on 3CX PBX,
At first I only created a trunk on elastix pbx and give it only the register string without any data on the peer details, the extension 103 on 3CX was registered with sucess and try to call any extension on 3CX PBX, I can´t
Afer this I give the trunk the data necesary on peer details and this time it was different I can call any extension on 3CX PBX.

After this I try to call any line outside 3CX PBX without sucess,

The only thing strange to me in the 3CX PBX is when I call from elastix any extension to 3CX any extension trought extension 103 the extension in 3CX ring but the extension 103 remain in green, as if this extension was free.

 

[jessydm]

I am having same issue....i can dial a 3cx extension but not 3cx trunk through asterisk..
Can anybody please share asterisk sip.conf section here ??

 

[leejor]

i can dial a 3cx extension but not 3cx trunk through asterisk

Perhaps you could provide a 3Cx log showing the successful extension call and an unsuccessful call to a trunk.

 

[brentrichburg]

Was there a successful solution to this thread?

 

[adam_devereaux]

The problem above is solved by entering the proper Peer information. Here is an example that worked for me with FreePBX:

Create an Extension in 3CX- make sure to set it to Use Phone Status.

Create your trunk in Asterisk- under Outgoing Settings type a useful name such as to-3cx. Enter the following in PEER details:

disallow=all
allow=ulaw
context=from-internal
type=friend
secret=(PASSWORD of EXTENSION)
nat=auto
insecure=very
host=(IP of 3CX)
username=(EXTENSION)
fromdomain=(IP of 3CX)
fromuser=(EXTENSION)
dtmfmode=auto

Not all of these may be necessary, but the most important are secret=, username=, fromdomain= and fromuser=.

Then under incoming settings leave the USER Details area blank.

I used the following Registration string:

extassword@ipof3cx/ext

so for example extension 3000 with a password of 123456 and the 3CX server having 192.168.20.10 would be:

3000:[email protected]/3000.

If you then create an extension on Asterisk and then create a trunk on 3cx pointing to those details you can create a full circle call path. If you play around with Outbound Dial Prefix and the Outbound Routes you can do a lot to create customized routes for call handling.

Adam Devereaux
Systems Engineer

Worksighted, Inc
http://www.worksighted.com
*IT Support. Redefined.

 

[cesargdmi]

for me not work, please any help me


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