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SIP-TRUNKING CONFIG FOR SAUDI ARABIA

Amod Joshi
Level 1
Level 1

‎06-19-2014 07:11 AM - edited ‎03-16-2019 11:09 PM

Hello,

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.

 

Thanks,

Labels:
0 Helpful
46 Replies 46

islam.kamal
Level 10
Level 10
In response to mohammadadeel

‎08-12-2014 06:12 AM

Mohamed

 

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/      

 

Thanks

please rate all useful information

Razeen Abdulla
Level 1
Level 1
In response to islam.kamal

‎11-08-2014 02:07 PM

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.

0 Helpful

islam.kamal
Level 10
Level 10
In response to mohammadadeel

‎08-12-2014 01:13 AM

Hello Mohamed

 

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

 

Thanks

please rate all useful information

Preview file
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islam.kamal
Level 10
Level 10
In response to islam.kamal

‎08-12-2014 01:14 AM

Hello

Kindly find the below

 

1-VGW connect to STC

interface  e0/1
no shut
ip address 172.29.55.90 255.255.255.250
duplex auto
speed auto

2-Route to SIP server  in STC
ip route 10.205.20.0 255.255.255.0 172.29.55.89

Note: now try to ping 172.29.55.89 , then ping 10.205.20.50


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 
 sip
  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 ipv4:10.201.20.50:5060
 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


Thanks

please rate all useful information

islam.kamal
Level 10
Level 10
In response to islam.kamal

‎08-12-2014 01:15 AM

Hello

Connectivity trouble shooting:
1.Be sure that your link is up and your IP “172.29.55.89” is defined in your side.
2.Be sure that you can reach the STC access side by pinging your gateway “172.29.55.89”.
4.Be sure to define the Sip server route toward the gateway by defining the route 
“Address 10.205.20.50* Mask 255.255.255.252 Next hub “172.29.27.157”
5.Be sure to reach the SIP server by Pinging the IP “10.205.20.50” 

 

Thanks

please rate all useful information

mohammadadeel
Level 1
Level 1
In response to islam.kamal

‎08-12-2014 02:35 AM

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.

 

Thanks

0 Helpful

islam.kamal
Level 10
Level 10
In response to mohammadadeel

‎08-12-2014 03:35 AM

Hi

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

Thanks

please rate all useful information

Preview file
1085 KB

mohammadadeel
Level 1
Level 1
In response to islam.kamal

‎08-12-2014 03:35 AM

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 10.10.5.2

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.

 

Thanks

0 Helpful

islam.kamal
Level 10
Level 10
In response to mohammadadeel

‎08-12-2014 03:56 AM

Hello

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 172.29.55.90. I have already updated on attached image.

 

Thanks 

please rate all useful information

mohammadadeel
Level 1
Level 1
In response to islam.kamal

‎08-12-2014 05:04 AM

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  ): 172.29.55.90
Destn SIP Req Addr:Port  : 10.205.20.50:0
Destn SIP Resp Addr:Port : 10.205.20.50:5060
Destination Name         : 10.205.20.50

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): 172.29.55.90
Source IP Port    (Media): 0
Destn  IP Address (Media): 10.205.20.50
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

 

0 Helpful

islam.kamal
Level 10
Level 10
In response to mohammadadeel

‎08-12-2014 06:25 AM

Hello

 

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
!
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 -
name:XCD123456

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

Go to SIP trunk- assign MRGL- reset

thanks

please rate all useful information

 

mohammadadeel
Level 1
Level 1
In response to islam.kamal

‎08-13-2014 01:37 AM

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)

 

Error

 

Eico-VGR(config)#dspfarm profile 1 transcode

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

% Unrecognized command

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

% Unrecognized command

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

  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.

 

Thanks

 

Preview file
4 KB
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islam.kamal
Level 10
Level 10
In response to mohammadadeel

‎08-13-2014 02:37 AM

Hello

do the below, please?.

router(config)# voice-card 0

dspfarm

dsp services dspfarm

 

Thanks

please rate all useful information

islam.kamal
Level 10
Level 10
In response to islam.kamal

‎08-13-2014 11:33 AM

Hello

Kindly check the routing  on your Core switch.

 

Thanks

please rate all useful information

mohammadadeel
Level 1
Level 1
In response to islam.kamal

‎08-13-2014 11:33 AM

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.

 

 

Thanks

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