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forward inbound GSM call from gsm gateway to Cisco CUCM

georgeotj
Level 1
Level 1

Hi,

 

I've created SIP Trunk between Yeastar TG-100 GSM Gateway and Cisco CUCM.

 

TG-100 ip address is = 172.68.9.41 

Cisco CUCM ip address is = 172.68.9.31 (extension number is 10xxx)

 

In TG-800, i create two call route, one for Cisco to GSM and other for GSM to Cisco.

 

VoIP Phones behind CUCM (10xxx) can make calls to GSM numbers (+62821xxxxxxx) through TG-100 successfully.

 

But when doing call outside office to GSM number at TG-100, i could get the Cisco CUCM tone but when dial the extension number, nothing happens, until call terminated.

 

Could anyone give me a hint what to check at Cisco CUCM to might be the culprit to this problem ?

 

Thanks.

1 Accepted Solution

Accepted Solutions

I finally solved it.

So, after i put the CSS as recommended by Nashaatmusa, i also create Route List, Route Group for this SIP Trunk and edit the Trunk Security Profile as attachment.

I do reset at the Trunk, SIP Trunk Security Profile, and suddenly it all works.

 

Thank you Nashaatmusa.

View solution in original post

7 Replies 7

Nashaatmusa
Level 1
Level 1
are your phones assigned partitions ?? ,, as the Trunk has no CSS !!

yes, i assigned partition to the voip phones.

do you mean that i had to config CSS (Calling Search Space) at the Cisco CUCM Trunk ?

(as attachment)

 

Thanks

Exactly, as the SIP trunk can't reach the phones with CSS assigned to it.

put CSS in place as picture , apply config , reset sip trunk if needed and give it a go

Ok, so i put the CSS in place, apply config. But still same result, i got Cisco CUCM tone but no other response until call ended.

I attached the Cisco CUCM trunk config i modified, and also below is the log from TG-100, maybe there are something in there, tbh i don't really understand the information on the log

 

Via: SIP/2.0/UDP 172.68.9.41:5060;branch=z9hG4bK5e464dcd;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.68.9.41>;tag=as7c358506
To: <sip:172.68.9.31>
Contact: <sip:Unknown@172.68.9.41>
Call-ID: 40e2099c0790557200d3fc187bd32320@172.68.9.41
CSeq: 102 OPTIONS
User-Agent: TG100
Date: Thu, 05 Jul 2018 16:22:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:172.68.9.31:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.68.9.41:5060;branch=z9hG4bK5e464dcd;rport
From: "Unknown" <sip:Unknown@172.68.9.41>;tag=as7c358506
To: <sip:172.68.9.31>;tag=1813280668
Date: Thu, 05 Jul 2018 16:22:07 GMT
Call-ID: 40e2099c0790557200d3fc187bd32320@172.68.9.41
Server: Cisco-CUCM12.0
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

 

 

Thanks

I finally solved it.

So, after i put the CSS as recommended by Nashaatmusa, i also create Route List, Route Group for this SIP Trunk and edit the Trunk Security Profile as attachment.

I do reset at the Trunk, SIP Trunk Security Profile, and suddenly it all works.

 

Thank you Nashaatmusa.

Good to know ! , best of luck

hello

i have the same probelm

why did you create route list and route group ??

the probelm for you was only on css si trunk ??

best regards