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SIP trunk Issues

Tasneem Fazal
Level 1
Level 1

Hi All, 

 

I am having issues in setting up SIP trunk with Service Provider, we are not receiving any calls (no Invite) and neither make any outbound calls, looks like its some interoperability  issue with telco which we are not been able to sort out. when I enable "debug ccsip messages".  I am continuously receiving debug messages below, I am not sure if its asking me to enable anything on my side more specifically?

 

Received:
OPTIONS sip:X.Y.178.74:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.110.139:5060;branch=z9hG4bKiybcyyeekcbwtnswydvycnbax;Role=3;Hpt=8e78_16;TRC=ffffffff-ffffffff;X-HwDim=4
Record-Route: <sip:X.X.110.139:5060;transport=udp;lr;Hpt=8e78_16;CxtId=4;TRC=ffffffff-ffffffff>
Call-ID: isbc6927772aa118x8aa7t1y48f26a8f25x6@B.5.103.ims.ooredoo.om
From: <sip:sobcf1.ims.ooredoo.om>;tag=414ya2f1
To: <sip:X.Y.178.74>
CSeq: 1 OPTIONS
Accept: application/sdp
Contact: <sip:X.X.110.139:5060;transport=udp;Dpt=7ba8_16;Hpt=8e78_16;CxtId=4;TRC=ffffffff-ffffffff>
Max-Forwards: 69
Content-Length: 0

*Sep 25 05:48:45.262: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:X.Y.178.74:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.119.75:5060;branch=z9hG4bK4cyewvdaickksyatvcdiiebsv;Role=3;Hpt=8e78_16;TRC=ffffffff-ffffffff;pth=0;X-HwDim=4
Call-ID: l4l4tkabiyyeweekyecatlnbxsttdvai@75.119.215.10
From: <sip:SBC@X.X.119.75>;tag=cswxnxaa
To: <sip:X.Y.178.74>
CSeq: 1 OPTIONS
Contact: <sip:X.X.119.75:36472;transport=udp;Hpt=8e78_16>;expires=65535
Accept: application/sdp
Max-Forwards: 70
Content-Length: 0

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

 

 

 

6 Replies 6

Deepak Kumar
VIP Alumni
VIP Alumni

Hi,

Share complete configuration.

Regards,
Deepak Kumar,
Don't forget to vote and accept the solution if this comment will help you!

Thanks Deepak for looking into this.
I didn't configure all the dial peers, just configured a few first testing. ISP told us to configure SIP option messages, what that mean? here is the config by the way.

voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
address-hiding
mode border-element license capacity 10
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
min-se 90 session-expires 300
header-passing
error-passthru
options-ping 60
no update-callerid
early-offer forced
midcall-signaling passthru
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
!
voice class server-group 1
ipv4 X.X.110.139 preference 1
ipv4 X.X.119.75 preference 2
description ISP-SIP-SERVERS
!
voice translation-rule 9
rule 1 /^9\(.*\)/ /\1/
!
voice translation-profile PSTN-OUT
translate called 9
!
!
dial-peer voice 1 voip
description *** Inbound dial-peer from CUCM ****
session protocol sipv2
incoming called-number 9T
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1.20
voice-class sip bind media source-interface GigabitEthernet0/0/1.20
dtmf-relay rtp-nte sip-info sip-notify
no vad
!
dial-peer voice 500 voip
description *** Outbound Dialpeer to CUCM ****
destination-pattern 225844..
session protocol sipv2
session target ipv4:X.Y.23.20
voice-class codec 1
voice-class sip options-ping 60
voice-class sip early-offer forced
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0/1.20
voice-class sip bind media source-interface GigabitEthernet0/0/1.20
dtmf-relay rtp-nte sip-info sip-notify
no vad
!
dial-peer voice 10 voip
description *** INTERNATIONAL CALLS ***
translation-profile outgoing PSTN-OUT
destination-pattern 900T
session protocol sipv2
session server-group 1
voice-class codec 1
voice-class sip options-ping 60
voice-class sip early-offer forced
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0/2
voice-class sip bind media source-interface GigabitEthernet0/0/2
dtmf-relay rtp-nte sip-info sip-notify
no vad
!
dial-peer voice 2 voip
description **** Inbound Dial-peer for receiving calls from SP *****
session protocol sipv2
session server-group 1
incoming called-number 225844..
voice-class codec 1
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/0/2
voice-class sip bind media source-interface GigabitEthernet0/0/2
dtmf-relay rtp-nte sip-info sip-notify
no vad
!
!
sip-ua
!

Hi Tasneem,

Options ping enables checking the SIP trunk status by sending heartbeats at regular intervals to the provider IP address. I have given a sample config below for your reference.

You need to configure a voice class profile for keepalives as below:

voice class sip-options-keepalive 1
description UDP Options consolidation
down-interval 49
up-interval 180
retry 7
transport udp

Apply this to all dial-peers pointing to provider SIP trunks. For example dial-peer 10 as below:

dial-peer voice 10 voip
voice-class sip options-keepalive profile 1

HTH
Rajan
Please rate all useful posts by clicking the star below and mark solutions as accepted wherever applicable

Thanks Rajan,
under dial-peer 10, I configured keepalive and option ping. Is that not sufficient?
moreover when when I enable "debug ccsip messages" , I am constantly receiving below messages which I am not sure indicate any underlying problem.

Received:
OPTIONS sip:10.240.178.74:5060 SIP/2.0

Via: SIP/2.0/UDP 10.215.119.75:5060;branch=z9hG4bKlklglflgf2kliohglfkf0grsh;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff;X-HwDim=4

Record-Route: <sip:10.215.119.75:5060;transport=udp;lr;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff>

Call-ID: isbc6f79f59axt3aab4f1b9ux762x1u14b6f@B.5.103.ims.ooredoo.om

From: <sip:sobcf1.ims.ooredoo.om>;tag=15t73axx

To: <sip:10.240.178.74>

CSeq: 1 OPTIONS

Accept: application/sdp

Contact: <sip:10.215.119.75:5060;transport=udp;Dpt=7ba8_16;Hpt=8e88_16;CxtId=4;TRC=ffffffff-ffffffff>

Max-Forwards: 69

Content-Length: 0




Sep 30 11:43:51.848: //18388/684C787787D9/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.215.119.75:5060;branch=z9hG4bKlklglflgf2kliohglfkf0grsh;Role=3;Hpt=8e88_16;TRC=ffffffff-ffffffff;X-HwDim=4

From: <sip:sobcf1.ims.ooredoo.om>;tag=15t73axx

To: <sip:10.240.178.74>;tag=120AC67-A19

Date: Mon, 30 Sep 2019 11:43:51 GMT

Call-ID: isbc6f79f59axt3aab4f1b9ux762x1u14b6f@B.5.103.ims.ooredoo.om

Server: Cisco-SIPGateway/IOS-16.6.4

CSeq: 1 OPTIONS

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Accept: application/sdp

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Content-Type: application/sdp

Content-Length: 377



v=0

o=CiscoSystemsSIP-GW-UserAgent 8086 5930 IN IP4 10.240.178.74

s=SIP Call

c=IN IP4 10.240.178.74

t=0 0

m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3

c=IN IP4 10.240.178.74

m=image 0 udptl t38

c=IN IP4 10.240.178.74

a=T38FaxVersion:0

a=T38MaxBitRate:9600

a=T38FaxRateManagement:transferredTCF

a=T38FaxMaxBuffer:200

a=T38FaxMaxDatagram:320

a=T38FaxUdpEC:t38UDPRedundancy

That should be fine but you should also this gateway/CUBE responding to those OPTIONS message from provider. Is that happening ?

Hi Rajan,
That is also my concern , i am not sure what should I enable to reply to those option messages. please advise?