Actually i m new to it with - Page 3

Amod Joshi · ‎06-19-2014

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,

islam.kamal · ‎08-13-2014

Hello

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

Thanks

please rate all useful information

mohammadadeel · ‎08-13-2014

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.

mohammadadeel · ‎08-12-2014

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/

islam.kamal · ‎08-12-2014

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

islam.kamal · ‎08-13-2014

Hello

We can do as the below

rule 1 /^.*$9...$/ /\1/

Thanks

please rate all useful information

duraimurugan_82 · ‎02-05-2015

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.

Suresh Subramanian · ‎02-05-2015

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

//Suresh Please rate all the useful posts.

Amod Joshi · ‎08-11-2014

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..

mohammadadeel · ‎08-12-2014

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.

midhun2064 · ‎08-12-2014

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

Razeen Abdulla · ‎11-08-2014

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

razeen@outlook.com

many thanks in advance.

florin.dovleac · ‎01-23-2020

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.

florin.dovleac · ‎01-23-2020

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

mohsin majeed · ‎08-13-2014

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 · ‎04-12-2015

Dear,

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

SIP-TRUNKING CONFIG FOR SAUDI ARABIA