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New Member



Does anyone know the sip trunk config for Saudi Arabia.

we have bought SIP trunk from Local ISTP, can we get  the configuration guide from ISTP.

what information will ISTP provide to us for the configuration of CUBE.


Any help will be highly appriciated.



Cisco Employee

Hi, Usually the ISP provides


Usually the ISP provides the following: 


Domain name


User ID

DID ranges

Number of Sessions

SBC and CPE IP address 

It depends on the provider, you'll need to check with them. 






Please rate helpful posts. 

Regards, Tere. If you find this post helpful, please rate! :)
New Member

Can you send me the basic

Can you send me the basic configuration steps for configuring sip trunk on cube

you can also check these URLs

you can also check these URLs.



//Suresh Please rate all the useful posts.
New Member

Dear all 

Dear all 

I have issues with stc sip line when i try to call from stc mobile  call not response  but when i try to from zain it is working perfectly 



Usually called number for calls originated from STC is appended with 011, while calls that originated from other ISP called number comes without 011.

Make sure incoming-called number/ destination number value are set properly on both incoming and outgoing dial-peers. I usually set it as: 0?1?1?44444XX, where 44444XX is an example range.

P.S. you can insert (?) by pressing ctrl+v then release keys and enter (?).

Thank you,


Sent from my mobile

New Member

make sure you have g711alaw

make sure you have g711alaw under the sip trunk configuration in CUCM

See attachment

Also try with enabling "Media Termination Point Required"

New Member

Hello, were you able to



were you able to configure your router, actually i am stuck with router configuration only.

i would appreciate if you can share your configuration file with me.

it would be a great help.


Thanks in Advance.


Hello Mehaomed Are you going

Hello Mehaomed 

Are you going to do integration with STC ?. For me i did many integration on KSA with STC all over KSA .



please rate all useful information

New Member

Hi Islam, Actually i m new to

Hi Islam,


Actually i m new to it with no prior experience, i would request you to share any running configuration file for cisco 2900 series router. i received configuration from stc:



Thanks in Advance.

Hello sure , i will provide



sure , i will provide you with a perfect configuration which already tested . I have only on question , did STC share with you any infformation about VLAN id to created?. just a question.



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New Member

no the above mentioned

no the above mentioned configuration is the only thing i received from STC no vlan ID.

New Member

Hi All,I have a similar

Hi All,

I have a similar situation with STC and no proper response from STC to troubleshoot. Was this issue resolved? I managed to get calls bothways but once AA transfers, the calls gets disconnected :(



MohamedAs per your DID


As per your DID 133459300-133459499, you have to  use range from 9XXX. because your range of your DID ends with 9XXX. If you use 3XX , you will lose the other range which starts with 4XX.



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New Member

please explain where do i

please explain where do i have to use this range? in router or in cucm.

Mohamed on CUCM , you will



on CUCM , you will configure the Phones who will need for DID from range 93XX to 94XX. On VG you have to change the translation rule 

rule 1 /^.*\(9...\)/  /\1/      



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New Member

dear islam..all my pings are

dear islam..all my pings are working, to the stc gw,  sip server and all. but now outgoing calls. i have 100 DID numbers. Can you provide a working config?

thanks in advance.

Hello Mohamed Kindly check

Hello Mohamed


Kindly check the attached draw which will give you the idea about connectivity between you and STC.



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HelloKindly find the below 1


Kindly find the below


1-VGW connect to STC

interface  e0/1
no shut
ip address
duplex auto
speed auto

2-Route to SIP server  in STC
ip route

Note: now try to ping , then ping

3-CUBE configuration on VGW

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
redirect ip2ip
fax protocol cisco 
  bind control source-interface eth0/1 
  bind media source-interface eth0/1

4-Outgoing call
dial-peer voice 100 voip
 description ** Outgoing Calls over SIP Trunk >> **
 destination-pattern .T
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target ipv4:
 dtmf-relay rtp-nte
 codec g711alaw
 no vad

5- incoming DID call
voice translation-rule 1
rule 1 /^.*\(3...\)/ /\1/      / This is example that your internal extensions 3XXX , which match last 4 digits for DIDs/

voice translation-profile DID
translate called 1

dial-peer voice 3000 voip
destination-pattern 3...
session protocol sipv2
sess target ipv4:x.x.x.x   /where x.x.x.x is the IP address for your CUCM/
 translation-profile incoming DID
 voice-class sip dtmf-relay force rtp-nte
 incoming called-number .
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad


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HelloConnectivity trouble


Connectivity trouble shooting:
1.Be sure that your link is up and your IP “” is defined in your side.
2.Be sure that you can reach the STC access side by pinging your gateway “”.
4.Be sure to define the Sip server route toward the gateway by defining the route 
“Address* Mask Next hub “”
5.Be sure to reach the SIP server by Pinging the IP “” 



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New Member

Thanks Islam, i have

Thanks Islam,


i have configured and checked ping is successfull, now what is the next step on cucm furthermore i had an old dial-peer configurations does it make any difference if i leave tham there or do i have to remove tham if so how do i remove tham from my router.



Hi1- For existing Dial-peers


1- For existing Dial-peers , if you are in  production , i suggest do not delete anything.

2- For SIP trunk configuration on CUCM. Go to CUCM- device-Trunk -add new - select SIP trunk - protocol SIP. Just see attached file for SIP trunk configuration

3- Go to CUCM -call routing - route /hunt- route group -add new 

name it , then select your SIP trunk which create on step 2.

4- Go to CUCM -call routing - route /hunt-- route list - add new - add RG which created on step 3.

5- Add new Route pattern. Go to CUCM -call routing - route /hunt-- route pattern- Add new 

Route pattern:8.XXXXXXXXXX  / I use  8 as i expected that you use 9 or you can use any outgoing code which will be different from your existing /

on Route List assign RL which created on step 4.

on disacrd digits : select predot 

6- Test


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New Member

Thanks a lot for your reply

Thanks a lot for your reply,

actually i created those dial peers while i was testing some code we are not yet live with cisco phones therefore i would like to remove all un-necessary configurations.

i have seen the attached file ip address you mentioned is for sip server and i used my router ip address that was

i will change it and follow your instructions.

by the way i need to give different access rights on calls like some employee will have local access only that is 133xxxxxxx.

some others will have mobile access like 05986488xx,

some will have landline facility only like 01122334455, 0123344555 etc.

some will have international calls like +92xxxxxxxxxx.

do i have to create different dial peers for all of the above mentioned access rights or is it a different story.

plz your help is highly appreciated.



Hellowe can configure PTs ,


we can configure PTs , and CSS to make priority for who can dial calls or not . Just we need to finish test that incoming and outgoing calls is very perfect then we can go for customization . For sure you have to put your SIP IP address I have already updated on attached image.



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New Member

My incomming call trace . Aug

My incomming call trace .


Aug 12 11:31:56.156: //36966/1975AD308C9D/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x31C7E2D0
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 506812001
Called Number            : 3459300
Source IP Address (Sig  ):
Destn SIP Req Addr:Port  :
Destn SIP Resp Addr:Port :
Destination Name         :

Aug 12 11:31:56.156: //36966/1975AD308C9D/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g729r8
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 18 (tx), 18 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media):
Source IP Port    (Media): 0
Destn  IP Address (Media):
Destn  IP Port    (Media): 54106
Orig Destn IP Address:Port (Media): [ - ]:0

Aug 12 11:31:56.156: //36966/1975AD308C9D/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 65
Disconnect Cause (SIP)   : 488


it gives me a dead tone. plz help


Hello 1- Did you test



1- Did you test incoming and outgoing ?. Kindly add the below:-

sccp local Ethernet0/0
sccp ccm X.X.X.X  identifier 1 priority 1 version 7.0+  /x.x.x.x the Ip address for CUCM/
sccp ccm group 1
 bind interface GigabitEthernet0/0
 associate ccm 1 priority 1
 associate profile 1 register XCD123456

dspfarm profile 1 transcode  
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 8
 associate application SCCP
no sh

Then add on CUCM- media resources - transcoder - select IOS enhanced -

Then - create MRG-asign the transcoder , after that create MRGL and assign MRG.

Go to SIP trunk- assign MRGL- reset


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New Member

Salam Islam,

Salam Islam,


if got the following errors while configuring  plz help.

(dspfarm profile 1 transcode  
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 8
 associate application SCCP
no sh)




Eico-VGR(config)#dspfarm profile 1 transcode

Eico-VGR(config-dspfarm-profile)#codec ?

% Unrecognized command

Eico-VGR(config-dspfarm-profile)#codec ?

% Unrecognized command


  associate    Associations this profile

  codec        The codec rate to be attempted for SCCP controlled connections

  description  Description about this profile

  exit         Exit from DSPFARM Profile configuration mode

  maximum      Configure maximum limit

  no           Negate a command or set its defaults

  rsvp         RSVP support for this profile

  shutdown     Disable or enable this profile

  stun         TRP  STUN support for this profile


I have attached my configuration file for your reference if  i am mastaking plz help.




Hellodo the below, please?


do the below, please?.

router(config)# voice-card 0


dsp services dspfarm



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HelloKindly check the routing


Kindly check the routing  on your Core switch.



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New Member

Thanks Islam for the promp

Thanks Islam for the promp reply,

it works, but for the session it only gave me the following 


Please configure the maximum sessions for this profile and retry
Eico-VGR(config-dspfarm-profile)#maximum sessions ?
  <1-2>  Number of sessions assigned to this profile

what shoud i select instead of 8 only 1 or 2 is available.