Need help to make outbound calls - CALL REJECTED

[Meejch]

Hello, my problem is that outbound calls aren't working. Inbound calls working fine. I have attached a logfile. Everytime the same message was shown: "call rejected". Please help me to make outgoing calls.

[Samir Doshi]

Call reject is from provider end. Please contact your provider. 

Best Regards
--
Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com

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On Wed, Feb 25, 2015 at 4:45 PM, Meejch <meej...@googlemail.com> wrote:

Hello, my problem is that outbound calls aren't working. Inbound calls working fine. I have attached a logfile. Everytime the same message was shown: "call rejected". Please help me to make outgoing calls.

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[Jon Morby]

2015-02-19 15:01:04.459479 [DEBUG] mod_sofia.c:407 sofia/default/70...@10.0.2.25 Overriding SIP cause 480 with 403 from the other leg

This shows an error with your outbound gateway .. make sure you’re registered and that the gateway accepts calls from your IP / etc

also check the dial string … do you have access to the logs on the gateway?  Can you see why it rejected the call?

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On 25 Feb 2015, at 11:15, Meejch <meej...@googlemail.com> wrote:

Hello, my problem is that outbound calls aren't working. Inbound calls working fine. I have attached a logfile. Everytime the same message was shown: "call rejected". Please help me to make outgoing calls.

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<outbound-call-log.txt>

[Meejch]

Thanks for the quick response.
I don't have access to the gateway log and i don't know why it rejected my calls. This is a german sipgate account. I have two seperate machines, the first one with a clean Freeswitch installation and the second one with Freeswith and ASTPP. The second machine was installed as described on your guide.
Both servers are using the same gateway and the same credentials for the sipgate account. On the the first server, freeswitch only, inbound and outbound calls are working fine.
On the second server, freeswitch + ASTPP, only inbound calls are working.

Does ASTPP change anything in the way it is handling outbound calls?

Which dialstring do you mean? The origination rate??

Am Mittwoch, 25. Februar 2015 13:00:35 UTC+1 schrieb Jon Morby (Fido):

2015-02-19 15:01:04.459479 [DEBUG] mod_sofia.c:407 sofia/def...@10.0.2.25 Overriding SIP cause 480 with 403 from the other leg

[Jon Morby]

Ok first thing … it isn’t "my guide" … I’m just a user / evaluating ASTPP just like you :) (and I have no affiliation with inextrix or astpp)

and I appreciate the docs are a bit “lean” which is why I thought I’d try and help out :)

Anyway … you need to have an outbound (termination) rate configured for your outbound calls / gateway, just as for your customers dialling out.  The difference between the customer rate and the gateway rate is your profit per call

If your clean freeswitch server is able to make calls out fine then I would suggest you add a gateway via the clean FS box onto your ASTPP box to help you debug in more detail

If that works then you’ve narrowed the problem down to a config issue between SipGate and your box .. if it doesn’t work then hopefully you’ll end up with more debug info

You can also add debug=1 to the sip_profile settings and turn on sip-trace in the sip_profile to see more of what’s happening also

worst case, use ngrep to see what’s happening at the SIP level and where the rejection is originating from / etc

www.fidonet.com - low cost domains, web hosting and virtual servers

www.fidotelecom.com - UK based ITSPA - holders of the ITSPA Quality Mark

[Meejch]

Hey !

Thanks for your advice !

I changed the gate configuration to the following: FS+ASTPP ---- gateway ----> FS (Clean installation) ---- gateway ----> SIPgate

so practically i am routing the call through the clean FS installation and its working... thank you VERY MUCH for you hint ! i would have never thought about this.

the question that remains: why does it not work when i configure the SIPgate as the direct gateway to ASTPP...  maybe you have an idea?
i will try again with another SIP provider, as soon i got one, but maybe it is working with the initial one.

thanks again ! you helped me a lot !
best regards from germany

[Jon Morby]

On 25 Feb 2015, at 13:12, Meejch <meej...@googlemail.com> wrote:

Hey !

Thanks for your advice !

I changed the gate configuration to the following: FS+ASTPP ---- gateway ----> FS (Clean installation) ---- gateway ----> SIPgate

so practically i am routing the call through the clean FS installation and its working... thank you VERY MUCH for you hint ! i would have never thought about this.

You’re welcome :)

the question that remains: why does it not work when i configure the SIPgate as the direct gateway to ASTPP...  maybe you have an idea?
i will try again with another SIP provider, as soon i got one, but maybe it is working with the initial one.

I’m guessing either a mistype of the credentials, or they are only allowing calls from certain IP addresses and your ASTPP box isn’t in their allow list ?

thanks again ! you helped me a lot ! 
best regards from germany