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Amod Joshi
Beginner

SIP-TRUNKING CONFIG FOR SAUDI ARABIA

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,

46 REPLIES 46

Hello

Kindly disable under voice-card 0 " codec complexity high".

 

Thanks

please rate all useful information

Hi Islam,

 

i am really confused at this stage for the following,

1. if i ping ip addresses from router for (172.29.55.90 and  172.29.55.80) it works fine, give me 100 % success.

2) if i ping the above mentioned ip address from 1 of the server where i have configured my vlan that is 10.10.5.x on of the network card, there is no ping for (172.29.55.90 and  172.29.55.80) but ping to my router gives me sucess on 10.10.5.2.

is it normal or what i think that there is no connectivity with STC, may be there is some issues with routing i am not sure.

unless i would be able to ping those ip addresses my incomming and outgoing calls will not work because even cucm (10.10.5.5) can not access (172.29.55.90 and 10.105.20.50)

 

your help is really appreciated.

 

Thanks a lot.

Thanks alot i will configure it and will get back to you if i stuck into someting furthermore, we have 3 digits extentions so i will use 3.. instead of 3... is it correct.

rule 1 /^.*\(3...\)/ /\1/      / This is example that your internal extensions 3XXX , which match last 4 digits for DIDs/

 

Hi

 

yes , correct and please do not forget to do rate and do correct answers when you finish and everything work as you expected.

 

Thanks

please rate all useful information

Hello

We can do as the below

rule 1 /^.*\(9...\)/ /\1/     

 

Thanks

please rate all useful information

 

Dear Islam,

 

Greetings. I am facing an issue with my call center. I am having problem only with outbound. i am getting address incomplete issue. I spoke to stc they told me the problem with dial plan. We tried all possibilities but still same, can you please help us.

 

awaiting for your response.

hello Durai, can you open a new thread with relevant call flow & debugs pls?

//Suresh Please rate all the useful posts.

Hi Mohammadadeel,

VoiceGateway -> STC Box -> STC Provider cloud

Make sure the both IP provided by the STC should be reachable.

Put  ip route 10.205.20.X 255.255.255.255 X.X.X.X

next thing, configure 

dial-peer voice 24 voip
 description To-Mobiles
 translation-profile outgoing To-SIP
 destination-pattern 05[0-9].......
 rtp payload-type cisco-codec-fax-ack 111
 rtp payload-type nte 97
 session protocol sipv2
 session target ipv4:10.205.20.X
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw
!

that's it no special configuration required.

 

If you need any further help, please contact me..

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. starting from scratch like what shoud be router ip address with port 0/0 that is going to communicate with cucm and on port 0/1 stc ip will be configured and binding between IP's.

 

may be i sound stupid but if you can provide any sample running config file that would be a great help and i will undrstand the whole id of configuration.

i received configuration from stc:

 

if i can get a configuration file it would be a great help.

Hi Mohammed,

Attached is a working configuration fro which have STC and GO SIP trunks.

 

Would you be able to give a brief regarding your setup. 

 

Which is you call agent ?? If you are using CUCM, how is it connecting to the voice gateway ??

 

Many thanks,

 

Midhun

 

 

can you send the working SIP trunking config for STC again? or send it at my email.

razeen@outlook.com

many thanks in advance.

Hello midhun,

Do you still have this working config? 

I'm having the same issue, router cisco 2911, interface giga0/0 from STC, 0/1 to edge switch Aruba and from Aruba to my Alcatel OXE (also i have routing enabled in switch, due to multiple vlans for ip telephones), i have ping to stc sip server but unable to make or receive any calls. STC tells me that they can ping me, they are sending option message fomr sip server to my pbx but nothing. So, please if you are able to share a configuration for the router, i would really much appreciate.

Hey Mohammed, you don't happen to still have the working config? Many thanks.

Aslamu alaikum, Mr Salam.

 

Great to see the discussion of you both the way to helping adeel. Me also from saudi arabia riyad.

Hi,

 

i am working on a customer site on cisco telephony. It is having SIP trunk with 100 DIDs. Outgoing call from site to the outside is working fine with no issue. But for incoming when i dial from my Cell phone to the company; cisco phone rings and when pick the phone no voice pass between and Cell phone play the continous ring even i pick the cisco phone for answer.

 

There are some debugs which are provided to me by ITSP having SIP/2.0 484 Address Incomplete issue. Can you please help your suggestion will be highly apprecialted

 

================================================================================

[No.           ] 1
[TimeStamp     ] 2014-08-13 14:49:44
[Direction     ] RECV
[Msg Name      ] INVITE
[Module No     ] 194
[Local Address ] 10.200.0.8:5060
[Remote Address] 10.208.16.8:5063
[Hex Msg       ] 49 4E 56 49 54 45 20 73 69 70 3A 30 31 31 32 38 33 35 34 32 ...


INVITE sip:0112835423@10.200.0.8;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.208.16.8:5063;branch=z9hG4bKehht5mgoypcpmm5hyyec1ysyp
Call-ID: gkmeyesteemm15eop3yypspiytsfmocc@SoftX3000
From: <sip:504595929@10.208.16.8;user=phone>;tag=m1k3mhck-CC-45
To: <sip:0112835423@10.200.0.8;user=phone>
CSeq: 1 INVITE
Contact: <sip:504595929@10.208.16.8:5063;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R010
Supported: 100rel
Max-Forwards: 70
Content-Length: 611
Content-Type: multipart/mixed;boundary=ssboundary

--ssboundary
Content-Length: 382
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 48206088 48206088 IN IP4 10.208.16.7
s=Sip Call
c=IN IP4 10.209.4.2
t=0 0
m=audio 32228 RTP/AVP 8 0 18 4 2 98 99 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

--ssboundary
Content-Length: 56
Content-Type: application/isup;version=itu-t92+

H
!8E2
TY)1X9x1:?:Cf
--ssboundary--

 

================================================================================

[No.           ] 2
[TimeStamp     ] 2014-08-13 14:49:43
[Direction     ] SEND
[Msg Name      ] INVITE
[Module No     ] 202
[Local Address ] 10.200.0.7:5069
[Remote Address] 10.200.20.235:5060
[Hex Msg       ] 49 4E 56 49 54 45 20 73 69 70 3A 30 31 31 32 38 33 35 34 32 ...


INVITE sip:0112835423@10.200.20.235;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.200.0.7:5069;branch=z9hG4bKhk4u7ueo77tbep7of7opcck4s
Call-ID: eobpesebthfc4fett7sou4ecdek2akae@SoftX3000
From: <sip:504595929@10.208.16.8;user=phone>;tag=hducfsbk-CC-36-TRC-703
To: <sip:0112835423@10.200.0.8;user=phone>
CSeq: 1 INVITE
Max-Forwards: 69
Contact: <sip:504595929@10.200.0.7:5069;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R010
Supported: 100rel
Content-Length: 381
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 27071045 27071045 IN IP4 10.200.0.7
s=Sip Call
c=IN IP4 10.209.4.2
t=0 0
m=audio 32228 RTP/AVP 8 0 18 4 2 98 99 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=fmtp:18 annexb=yes

 

================================================================================

[No.           ] 3
[TimeStamp     ] 2014-08-13 14:49:43
[Direction     ] RECV
[Msg Name      ] 100
[Module No     ] 202
[Local Address ] 10.200.0.7:5069
[Remote Address] 10.200.20.235:5060
[Hex Msg       ] 53 49 50 2F 32 2E 30 20 31 30 30 20 54 72 79 69 6E 67 0D 0A ...


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.200.0.7:5069;branch=z9hG4bKhk4u7ueo77tbep7of7opcck4s
Call-ID: eobpesebthfc4fett7sou4ecdek2akae@SoftX3000
From: <sip:504595929@10.208.16.8;user=phone>;tag=hducfsbk-CC-36-TRC-703
To: <sip:0112835423@10.200.0.8;user=phone>
CSeq: 1 INVITE
Content-Length: 0

 


================================================================================

[No.           ] 4
[TimeStamp     ] 2014-08-13 14:49:44
[Direction     ] SEND
[Msg Name      ] 100
[Module No     ] 194
[Local Address ] 10.200.0.8:5060
[Remote Address] 10.208.16.8:5063
[Hex Msg       ] 53 49 50 2F 32 2E 30 20 31 30 30 20 54 72 79 69 6E 67 0D 0A ...


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.208.16.8:5063;branch=z9hG4bKehht5mgoypcpmm5hyyec1ysyp
Call-ID: gkmeyesteemm15eop3yypspiytsfmocc@SoftX3000
From: <sip:504595929@10.208.16.8;user=phone>;tag=m1k3mhck-CC-45
To: <sip:0112835423@10.200.0.8;user=phone>
CSeq: 1 INVITE
Content-Length: 0

 


================================================================================

[No.           ] 5
[TimeStamp     ] 2014-08-13 14:49:43
[Direction     ] RECV
[Msg Name      ] 484
[Module No     ] 202
[Local Address ] 10.200.0.7:5069
[Remote Address] 10.200.20.235:5060
[Hex Msg       ] 53 49 50 2F 32 2E 30 20 34 38 34 20 41 64 64 72 65 73 73 20 ...


SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.200.0.7:5069;branch=z9hG4bKhk4u7ueo77tbep7of7opcck4s
Record-Route: <sip:10.200.20.235:5060;transport=udp;lr>
Call-ID: eobpesebthfc4fett7sou4ecdek2akae@SoftX3000
From: <sip:504595929@10.208.16.8;user=phone>;tag=hducfsbk-CC-36-TRC-703
To: <sip:0112835423@10.200.0.8;user=phone>;tag=sbc080787C44FE0-1AC9
CSeq: 1 INVITE
Date: Wed, 13 Aug 2014 11:49:44 GMT
Allow-Events: kpml,telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=28
Content-Length: 0

 


================================================================================

[No.           ] 6
[TimeStamp     ] 2014-08-13 14:49:43
[Direction     ] SEND
[Msg Name      ] ACK
[Module No     ] 202
[Local Address ] 10.200.0.7:5069
[Remote Address] 10.200.20.235:5060
[Hex Msg       ] 41 43 4B 20 73 69 70 3A 30 31 31 32 38 33 35 34 32 33 40 31 ...


ACK sip:0112835423@10.200.20.235;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.200.0.7:5069;branch=z9hG4bKhk4u7ueo77tbep7of7opcck4s
Call-ID: eobpesebthfc4fett7sou4ecdek2akae@SoftX3000
From: <sip:504595929@10.208.16.8;user=phone>;tag=hducfsbk-CC-36-TRC-703
To: <sip:0112835423@10.200.0.8;user=phone>;tag=sbc080787C44FE0-1AC9
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

 


================================================================================

[No.           ] 7
[TimeStamp     ] 2014-08-13 14:49:44
[Direction     ] SEND
[Msg Name      ] 484
[Module No     ] 194
[Local Address ] 10.200.0.8:5060
[Remote Address] 10.208.16.8:5063
[Hex Msg       ] 53 49 50 2F 32 2E 30 20 34 38 34 20 41 64 64 72 65 73 73 20 ...


SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.208.16.8:5063;branch=z9hG4bKehht5mgoypcpmm5hyyec1ysyp
Call-ID: gkmeyesteemm15eop3yypspiytsfmocc@SoftX3000
From: <sip:504595929@10.208.16.8;user=phone>;tag=m1k3mhck-CC-45
To: <sip:0112835423@10.200.0.8;user=phone>;tag=hh2ketup
CSeq: 1 INVITE
Reason: Q.850;cause=28;text="address incomplete"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 6
Content-Type: application/isup;version=itu-t92+

 


================================================================================

[No.           ] 8
[TimeStamp     ] 2014-08-13 14:49:44
[Direction     ] RECV
[Msg Name      ] ACK
[Module No     ] 194
[Local Address ] 10.200.0.8:5060
[Remote Address] 10.208.16.8:5063
[Hex Msg       ] 41 43 4B 20 73 69 70 3A 30 31 31 32 38 33 35 34 32 33 40 31 ...


ACK sip:0112835423@10.200.0.8;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.208.16.8:5063;branch=z9hG4bKehht5mgoypcpmm5hyyec1ysyp
Call-ID: gkmeyesteemm15eop3yypspiytsfmocc@SoftX3000
From: <sip:504595929@10.208.16.8;user=phone>;tag=m1k3mhck-CC-45
To: <sip:0112835423@10.200.0.8;user=phone>;tag=hh2ketup
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

 

 

 

 

=======================================

Here is my configuration of the cisco Gateway

 


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 h450.2
 no supplementary-service h450.3
 no supplementary-service h225-notify cid-update
 redirect ip2ip
 fax protocol pass-through g711alaw
 h323
 sip
  rel1xx disable
  header-passing
  registrar server expires max 3600 min 3500
  transport switch udp tcp
  redirect contact order best-match
  midcall-signaling passthru
!


voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  registrar server

----------------------------------------------------------

!
voice translation-rule 1
 rule 1 /^11/ /0/
 rule 2 /^2/ /02/
 rule 3 /^3/ /03/
 rule 4 /^4/ /04/
 rule 5 /^5/ /005/
 rule 6 /^6/ /06/
 rule 7 /^7/ /07/
 rule 8 /^1/ /001/
!
voice translation-rule 2
 rule 1 /^0\(\)/ /\1/
!
voice translation-rule 10
 rule 1 /^11/ /0/
 rule 2 /^12/ /0012/
 rule 3 /^13/ /0013/
 rule 4 /^14/ /0014/
 rule 5 /^5/ /005/
 rule 6 /^16/ /0016/
 rule 7 /^17/ /0017/
 rule 8 /^18/ /0018/
!
voice translation-rule 20
 rule 1 /^54..$/ /283\0/
!
voice translation-rule 120
 rule 1 /^89..$/ /243\0/
!
!
voice translation-profile INCO_SIP
 translate calling 10
!
voice translation-profile PSTN-IN
 translate calling 1
!
voice translation-profile PSTN-OUT
 translate called 2
!
voice translation-profile SIP
 translate calling 20
 translate called 10
!
voice translation-profile SIP2
 translate calling 120
 translate called 10

 

 

 

dial-peer voice 14 voip
 description Outgoing
 translation-profile outgoing SIP
 destination-pattern .T
 rtp payload-type cisco-codec-fax-ack 111
 rtp payload-type nte 97
 session protocol sipv2
 session target ipv4:10.200.7.157:5060
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
dial-peer voice 17 voip
 description incoming From STC Server to CUCM
 preference 1
 destination-pattern ^28354..$
 session target ipv4:172.16.200.13
 voice-class codec 1
 dtmf-relay h245-alphanumeric
 no vad
!
dial-peer voice 18 voip
 description incoming From STC Server to CUCM 
 destination-pattern ^28354..$
 session target ipv4:172.16.200.20
 voice-class codec 1
 dtmf-relay h245-alphanumeric 
 no vad


dial-peer voice 10 voip
 translation-profile incoming INCO_SIP
 session protocol sipv2
 session target sip-server
 incoming called-number ^28354..$
 dtmf-relay rtp-nte
 codec g711alaw
 no vad

 

 

 

 

 

 

 

 

Eldho E Kuriakose
Participant

Dear,

From my range 8007490000 is not working. Can anybody try dialing this from their SIP trunk and have a try?

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