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Cisco 2851 w/ CME 8.x & SIP Trunk via Vitelity

Unanswered Question
Dec 15th, 2013
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I have a Cisco 2851 loaded with DSPs and it has an internet connection to support the SIP trunk to Vitelity. Below are the appropriate sections of my config for this setup. I am running CME in both SCCP and SIP which is why i have some SIP bind to interfaces.

Here are the things i would like to get functioning...

- Incoming calls working properly... Currently i get a slow busy when I try to call my DID at Vitelity. Outgoing works correctly.

- Outgoing callerID setup correctly. How do i specify a static # and how do i specify 123456XXXX like CM works.

- Currently when i dial out i don't have to dial 9... 9 doesn't work to dial out, just regular dialing.


---------


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 pass-through g711ulaw

h323

  h450 h450-2 timeout T1 1000

  h450 h450-3 timeout T1 1000

  h225 display-ie ccm-compatible

modem passthrough nse codec g711ulaw

sip

  bind control source-interface GigabitEthernet0/0

  bind media source-interface GigabitEthernet0/0

  registrar server expires max 600 min 60

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

!

voice class h323 1

  h225 timeout tcp establish 3

  h225 timeout setup 2

  call start fast

  telephony-service ccm-compatible

  ccm-compatible


dial-peer voice 9 voip

destination-pattern .T

session protocol sipv2

session target sip-server

session transport udp

incoming called-number 9T

voice-class codec 1

dtmf-relay rtp-nte

!

!

sip-ua

credentials username *** password 7 *** realm asterisk

authentication username *** password 7 *** realm asterisk

no remote-party-id

retry invite 2

retry register 2

registrar dns:sip32.vitelity.net expires 3600

sip-server dns:outbound.vitelity.net

host-registrar


!

telephony-service

moh-file-buffer 1024

authentication credential rtradm rtradm

max-ephones 100

max-dn 250

ip source-address 10.x.x.9 port 2000

timeouts interdigit 5

system message Corp Office

url services http://10.x.x.8/voiceview/common/login.do

url authentication http://10.x.x.9/CCMCIP/authenticate.asp

cnf-file location flash:

load 7937 apps37sccp.1-4-5-7.bin

load 7942 SCCP42.9-3-1SR2-1S.loads

load 7945 SCCP45.9-3-1SR2-1S.loads

load 7962 SCCP42.9-3-1SR2-1S.loads

load 7965 SCCP45.9-3-1SR2-1S.loads

keepalive 10 auxiliary 10

voicemail 6500

max-conferences 16 gain -6

moh music-on-hold.au

multicast moh 239.1.1.1 port 16384 route 10.x.x.9

web admin system name rtradm password rtradm

transfer-system full-consult

secondary-dialtone 9

create cnf-files version-stamp 7960 Dec 15 2013 19:26:37

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Ayodeji Okanlawon Mon, 12/16/2013 - 02:02
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Lets start by looking at the incoming call issues..


Please do a test cal and sned us


debug ccsip messages

debug voip ccapi inout



Please rate all useful posts


"opportunity is a haughty goddess who waste no time with those who are unprepared"

gcrawford2005 Sun, 12/29/2013 - 21:16
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Below is an output of an inbound call...


----------

Dec 30 04:48:32.439: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:[email protected]:56342 SIP/2.0


Via: SIP/2.0/UDP 66.241.99.90:5060;branch=z9hG4bK209f473e;rport


From: "+1OutsideCaller" ;tag=as7e32b698


To:


Contact:


Call-ID: [email protected]


CSeq: 102 INVITE


User-Agent: packetrino


Max-Forwards: 70


Date: Mon, 30 Dec 2013 04:48:32 GMT


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


Supported: replaces


Content-Type: application/sdp


Content-Length: 334



v=0


o=root 22208 22208 IN IP4 66.241.99.90


s=session


c=IN IP4 66.241.99.90


t=0 0


m=audio 17164 RTP/AVP 0 8 3 18 101


a=rtpmap:0 PCMU/8000


a=rtpmap:8 PCMA/8000


a=rtpmap:3 GSM/8000


a=rtpmap:18 G729/8000


a=fmtp:18 annexb=no


a=rtpmap:101 telephone-event/8000


a=fmtp:101 0-16


a=silenceSupp:off - - - -


a=ptime:20


a=sendrecv



Dec 30 04:48:32.451: //93/77FD6BB28038/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying


Via: SIP/2.0/UDP 66.241.99.90:5060;branch=z9hG4bK209f473e;rport


From: "+1OutsideCaller" ;tag=as7e32b698


To:


Date: Mon, 30 Dec 2013 04:48:32 GMT


Call-ID: [email protected]


CSeq: 102 INVITE


Allow-Events: telephone-event


Server: Cisco-SIPGateway/IOS-12.x


Content-Length: 0




Dec 30 04:48:32.455: //93/77FD6BB28038/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 403 Forbidden


Via: SIP/2.0/UDP 66.241.99.90:5060;branch=z9hG4bK209f473e;rport


From: "+1OutsideCaller" ;tag=as7e32b698


To: ;tag=3126AC-AD5


Date: Mon, 30 Dec 2013 04:48:32 GMT


Call-ID: [email protected]


CSeq: 102 INVITE


Allow-Events: telephone-event


Server: Cisco-SIPGateway/IOS-12.x


Reason: Q.850;cause=21


Content-Length: 0




Dec 30 04:48:32.503: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:[email protected]:56342 SIP/2.0


Via: SIP/2.0/UDP 66.241.99.90:5060;branch=z9hG4bK209f473e;rport


From: "+1OutsideCaller" ;tag=as7e32b698


To: ;tag=3126AC-AD5


Contact:


Call-ID: [email protected]


CSeq: 102 ACK


User-Agent: packetrino


Max-Forwards: 70


Content-Length: 0




Dec 30 04:48:32.935: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

REGISTER sip:sip32.vitelity.net:5060 SIP/2.0


Via: SIP/2.0/UDP MYinsideIP:5060;branch=z9hG4bK82388


From: [email protected]>;tag=31288C-13FC


To: [email protected]>


Date: Mon, 30 Dec 2013 04:48:32 GMT


Call-ID: 999E3908-705211E3-8004C118-6EF80127


User-Agent: Cisco-SIPGateway/IOS-12.x


Max-Forwards: 70


Timestamp: 1388378912


CSeq: 18 REGISTER


Contact:


Expires:  3600


Supported: path


Content-Length: 0




Dec 30 04:48:33.435: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

REGISTER sip:sip32.vitelity.net:5060 SIP/2.0


Via: SIP/2.0/UDP MYinsideIP:5060;branch=z9hG4bK82388


From: [email protected]>;tag=31288C-13FC


To: [email protected]>


Date: Mon, 30 Dec 2013 04:48:33 GMT


Call-ID: 999E3908-705211E3-8004C118-6EF80127


User-Agent: Cisco-SIPGateway/IOS-12.x


Max-Forwards: 70


Timestamp: 1388378913


CSeq: 18 REGISTER


Contact:


Expires:  3600


Supported: path


Content-Length: 0




Dec 30 04:48:34.435: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

REGISTER sip:sip32.vitelity.net:5060 SIP/2.0


Via: SIP/2.0/UDP MYinsideIP:5060;branch=z9hG4bK82388


From: [email protected]>;tag=31288C-13FC


To: [email protected]>


Date: Mon, 30 Dec 2013 04:48:34 GMT


Call-ID: 999E3908-705211E3-8004C118-6EF80127


User-Agent: Cisco-SIPGateway/IOS-12.x


Max-Forwards: 70


Timestamp: 1388378914


CSeq: 18 REGISTER


Contact:


Expires:  3600


Supported: path


Content-Length: 0




Dec 30 04:48:34.551: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:[email protected]:56342 SIP/2.0


Via: SIP/2.0/UDP 66.241.99.90:5060;branch=z9hG4bK66b800c4;rport


From: "+1OutsideCaller" ;tag=as3fb078c9


To:


Contact:


Call-ID: [email protected]


CSeq: 102 INVITE


User-Agent: packetrino


Max-Forwards: 70


Date: Mon, 30 Dec 2013 04:48:34 GMT


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


Supported: replaces


Content-Type: application/sdp


Content-Length: 334



v=0


o=root 22208 22208 IN IP4 66.241.99.90


s=session


c=IN IP4 66.241.99.90


t=0 0


m=audio 15832 RTP/AVP 0 8 3 18 101


a=rtpmap:0 PCMU/8000


a=rtpmap:8 PCMA/8000


a=rtpmap:3 GSM/8000


a=rtpmap:18 G729/8000


a=fmtp:18 annexb=no


a=rtpmap:101 telephone-event/8000


a=fmtp:101 0-16


a=silenceSupp:off - - - -


a=ptime:20


a=sendrecv



Dec 30 04:48:34.563: //95/793F13E4803D/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying


Via: SIP/2.0/UDP 66.241.99.90:5060;branch=z9hG4bK66b800c4;rport


From: "+1OutsideCaller" ;tag=as3fb078c9


To:


Date: Mon, 30 Dec 2013 04:48:34 GMT


Call-ID: [email protected]


CSeq: 102 INVITE


Allow-Events: telephone-event


Server: Cisco-SIPGateway/IOS-12.x


Content-Length: 0




Dec 30 04:48:34.563: //95/793F13E4803D/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 403 Forbidden


Via: SIP/2.0/UDP 66.241.99.90:5060;branch=z9hG4bK66b800c4;rport


From: "+1OutsideCaller" ;tag=as3fb078c9


To: ;tag=312EE4-2A1


Date: Mon, 30 Dec 2013 04:48:34 GMT


Call-ID: [email protected]


CSeq: 102 INVITE


Allow-Events: telephone-event


Server: Cisco-SIPGateway/IOS-12.x


Reason: Q.850;cause=21


Content-Length: 0




Dec 30 04:48:34.615: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:[email protected]:56342 SIP/2.0


Via: SIP/2.0/UDP 66.241.99.90:5060;branch=z9hG4bK66b800c4;rport


From: "+1OutsideCaller" ;tag=as3fb078c9


To: ;tag=312EE4-2A1


Contact:


Call-ID: [email protected]


CSeq: 102 ACK


User-Agent: packetrino


Max-Forwards: 70


Content-Length: 0




Router#

Suresh Subramanian Sun, 12/29/2013 - 23:40
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you are receiving INVITE message from this ip: 66.241.99.90. have you configured this ip in cme to accept the sip connection?

Ayodeji Okanlawon Mon, 12/30/2013 - 02:41
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    2017 IP Telephony

Like Suresh said, this looks like toll fraud prevention. You need to add the ips of your sip provider to your ip addres trusted list


voice service voip

ip address trusted list

ipv4 66.241.99.90 X.X.X.X where x.x.x.x = subnet mask of the ip


Please rate all useful posts


"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

gcrawford2005 Mon, 12/30/2013 - 07:54
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I added the below to cover my basis since my SIP provider has an IP range and I am using DNS names instead of IPs for the connection. Still getting a slow busy when calling the outside number. From my SIP provider i get a message saying... "We received 'CONGESTION' when attempting to route the call to your server or device."  At the bottom is a new debug log of whats happening. Just a side note, i do have an ASA 5505 between the internet and my 28xx router.


voice service voip

ip address trusted list

  ipv4 66.241.96.0 255.255.240.0

  ipv4 66.241.99.90 255.255.255.255


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

Router#

Dec 30 15:43:29.562: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:[email protected]:54766 SIP/2.0


Via: SIP/2.0/UDP MYsipProvider:5060;branch=z9hG4bK1b517ef6;rport


From: "+1ExternalCallerNumber" ;tag=as186e2b7c


To:


Contact:


Call-ID: [email protected]


CSeq: 102 INVITE


User-Agent: packetrino


Max-Forwards: 70


Date: Mon, 30 Dec 2013 15:43:29 GMT


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


Supported: replaces


Content-Type: application/sdp


Content-Length: 334




v=0


o=root 22208 22208 IN IP4 MYsipProvider


s=session


c=IN IP4 MYsipProvider


t=0 0


m=audio 19262 RTP/AVP 0 8 3 18 101


a=rtpmap:0 PCMU/8000


a=rtpmap:8 PCMA/8000


a=rtpmap:3 GSM/8000


a=rtpmap:18 G729/8000


a=fmtp:18 annexb=no


a=rtpmap:101 telephone-event/8000


a=fmtp:101 0-16


a=silenceSupp:off - - - -


a=ptime:20


a=sendrecv



Dec 30 15:43:29.566: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 400 Bad Request - 'Invalid Host'


Via: SIP/2.0/UDP MYsipProvider:5060;branch=z9hG4bK1b517ef6;rport


From: "+1ExternalCallerNumber" ;tag=as186e2b7c


To: ;tag=1F4309C-23DB


Date: Mon, 30 Dec 2013 15:43:29 GMT


Call-ID: [email protected]


CSeq: 102 INVITE


Allow-Events: telephone-event


Reason: Q.850;cause=100


Server: Cisco-SIPGateway/IOS-12.x


Content-Length: 0





Dec 30 15:43:29.598: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:[email protected]:54766 SIP/2.0


Via: SIP/2.0/UDP MYsipProvider:5060;branch=z9hG4bK1b517ef6;rport


From: "+1ExternalCallerNumber" ;tag=as186e2b7c


To: ;tag=1F4309C-23DB


Contact:


Call-ID: [email protected]


CSeq: 102 ACK


User-Agent: packetrino


Max-Forwards: 70


Content-Length: 0





Router#

Suresh Subramanian Mon, 12/30/2013 - 10:19
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When the gateway processes an initial INVITE, a determination is made  whether or not the host portion is in IPv4 format or a domain name.

If  the host portion is an IP address, its IP address is compared with the  interfaces on the gateway.

If a match is found, the INVITE is processed  as normal.

If there is not a match, the gateway sends a 400 Bad Request –  `Invalid IP Address' message


could you please verify 'MYexternalIP' is configured? are you doing any NAT?

gcrawford2005 Mon, 12/30/2013 - 13:54
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MYexternalIP is my ISPs IP assigned to the WAN interface of my ASA5505. The 28xx sits on the LAN side of my ASA on a 10.x.x.99 IP. The ASA is doing NAT between the outside and inside.  Does that answer the question? On my SIP providers side there is an option to enable NAT and that is currently turned on and believe it gets set when the box registers itself.

gcrawford2005 Fri, 01/03/2014 - 15:01
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Did the above make sense to anyone? If anyone is availible over the weekend i'd really like to resolve these issues. I'd be willing to compensate someone for their time, just PM me or reply.


Thanks,

Ayodeji Okanlawon Sat, 01/04/2014 - 03:58
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    2017 IP Telephony

From the logs, the external IP was not natted to the internal IP of your gateway. The ASA sent the packet as it is, hence the gateway is rejecting the call as Suresh already mentioned. You need to ensure that the externalIP is Natted to gateway's IP before the INVITE is sent to it..



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"The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

gcrawford2005 Sun, 01/05/2014 - 11:13
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So i'm not quite following this... I have my ASA 5505 between the Internet and Inside doing nat from Public IP -> Internal IP range. I have a single static IP on the outside, which the ASA is using. Other computers on my Internet Vlan get to the internet just fine with the nat in place for 10.0.0.x to translated to my ASA's outside IP. Do I need to setup NAT on my Cisco 2811 Router thats doing SIP to translate its internet IP to the external IP? If this is the case will that affect any Lan traffic between the router and the phones? or am i completely missing something?

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