Solved: Re: Callcentric SIP Trunk (ITSP --> 2811 CUBE --> CUCM 8.6 - Page 2

whatuusay1 · ‎02-14-2014

I have a SIP trunk from call centric that goes into my lab gear - they appear to be a good sip service due to cost but I'm having some trouble getting calls to route correctly. The call flow is Callcentric.com ITSP (SIP) --> 2811 (acting as cube) -->SIP Trunk --> CUCM 8.6. Phones are registered to CUCM.

I have the sip trunk registered and calls come in to the router (I see them in ccsip message/call debugs) The 2811 running 15.1(4)M7). Callcentric sends the username of the customer in the sip Invite instead of the called number, the called number is in the TO field. I have several DID’s from Callcentric (18452055544, 18452055545, 18452055546) for my lab. There are a few configs on here for CME where the customer number (17772253754) is simply translated to their phone DN - which is fine if you only have 1 DN with callcentric but more than 1 and thats not feasible since every inbound did will be matched to that 17772253754 translation/phone dn.

I’m using the a guide from http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/ using the Copy function as described http://www.cisco.com/c/en/us/products/collateral/ios-nx-os-software/ios-software-release-15-1-3-t/product_bulletin_c25-635704.html

I haven’t been able to find anything where they actually explain all the header fields so Its mostly trial and error.. so far mostly error. I think I’m close.. but who knows. Any assistance would be greatly appreciated

voice class sip-profiles 1

request INVITE peer-header sip TO copy ".sip:(.*)@." u01

request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"

CUCM (single/pub)- 192.168.1.200

2811 acting as cube - 192.168.1.203

Calling Number - 18165297500

Called Number - 18452055544

vrtr1#show sip register status

Line peer expires(sec) registered P-Associ-URI

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

17772253754 -1 20 yes

vrtr1#

The Call Setup Information is:

Call Control Block (CCB) : 0x49646C28

State of The Call : STATE_DEAD

TCP Sockets Used : NO

Calling Number : 18165297500

Called Number : 17772253754 (my customer number not called number)

Source IP Address (Sig ): 192.168.1.203 (my 2811 router)

Destn SIP Req Addr:Port : 204.11.192.159:5080

Destn SIP Resp Addr:Port : 204.11.192.159:5080

Destination Name : 204.11.192.159

Feb 14 11:20:53.303: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:17772253754@192.168.1.203:5060 SIP/2.0

v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74

f: <sip:18165297500@66.193.176.35>;tag=3601387252-874282

t: <sip:18452055544@ss.callcentric.com>

i: 2917398-3601387252-874253@msw2.telengy.net

CSeq: 1 INVITE

Max-Forwards: 8

m: <sip:ca7fe50c9cebe12327fe0d63c5962a3e@204.11.192.159:5080;transport=udp>

Supported: timer

c: application/sdp

l: 350

v=0

o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.159

s=sip call

c=IN IP4 204.11.192.159

t=0 0

m=audio 61094 RTP/AVP 18 0 8 101

a=fmtp:18 annexb=no

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=ptime:20

a=sendrecv

a=silenceSupp:off - - - -

a=setup:actpass

Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74

From: <sip:18165297500@66.193.176.35>;tag=3601387252-874282

To: <sip:18452055544@ss.callcentric.com>

Date: Fri, 14 Feb 2014 17:20:53 GMT

Call-ID: 2917398-3601387252-874253@msw2.telengy.net

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74

From: <sip:18165297500@66.193.176.35>;tag=3601387252-874282

To: <sip:18452055544@ss.callcentric.com>;tag=35399D8-63

Date: Fri, 14 Feb 2014 17:20:53 GMT

Call-ID: 2917398-3601387252-874253@msw2.telengy.net

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=1

Content-Length: 0

Feb 14 11:20:53.419: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:17772253754@192.168.1.203:5060 SIP/2.0

v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74

f: <sip:18165297500@66.193.176.35>;tag=3601387252-874282

t: <sip:18452055544@ss.callcentric.com>;tag=35399D8-63

i: 2917398-3601387252-874253@msw2.telengy.net

CSeq: 1 ACK

Max-Forwards: 10

l: 0

u all

Feb 14 11:20:57.067: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:18452055544;cic=0288;rn=6465471001;npdi@alpha14.callcentric.com:5070 SIP/2.0

v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-6bceae47efe9f53b4234698a32ac8beb

f: <sip:18165297500@66.193.176.35>;tag=3601387252-874282

t: <sip:18452055544@ss.callcentric.com>;tag=35399D8-63

i: 2917398-3601387252-874253@msw2.telengy.net

CSeq: 1 ACK

Max-Forwards: 8

l: 0

************************** Running Config **************************

sh run
vrtr1#sh running-config
Building configuration...

Current configuration : 4189 bytes
!
! Last configuration change at 00:34:03 CST Fri Feb 14 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
version 15.1
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
no service password-encryption
!
hostname vrtr1
!
boot-start-marker
boot system flash:
boot system flash flash:c2800nm-ipvoicek9-mz.151-4.M7.bin
boot-end-marker
!
!
card type t1 0 0
logging buffered 4096 notifications
enable password cisco
!
no aaa new-model
memory-size iomem 5
clock timezone CST -6 0
clock summer-time CST recurring
no network-clock-participate wic 0
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
!
ip name-server 192.168.1.9
no ipv6 cef
multilink bundle-name authenticated
!
!
!
!
!
!
!
voice service voip
ip address trusted list
ipv4 192.168.1.0 255.255.255.0
ipv4 204.11.192.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server expires max 1800 min 1800
localhost dns:callcentric.com
outbound-proxy dns:callcentric.com
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
!
voice class sip-profiles 1
request INVITE peer-header sip TO copy ".sip:(.*)@." u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
!
!
!
!
!
voice-card 0
!
crypto pki token default removal timeout 0
!
!
!
!
license udi pid CISCO2811 sn FTX1133A4QR
!
!
controller T1 0/0/0
cablelength long 0db
!
!
!
!
!
interface FastEthernet0/0
description ** LAN **
ip address 192.168.1.203 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.1.203
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 192.168.1.1
!
!
snmp mib persist circuit
!
!
control-plane
!
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 192.168.1.200
ccm-manager config
!
mgcp
mgcp call-agent 192.168.1.200 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface FastEthernet0/0
mgcp bind media source-interface FastEthernet0/0
!
mgcp profile default
!
!
dial-peer voice 999100 pots
service mgcpapp
port 0/1/0
!
dial-peer voice 999101 pots
service mgcpapp
port 0/1/1
!
dial-peer voice 999102 pots
service mgcpapp
port 0/1/2
!
dial-peer voice 999103 pots
service mgcpapp
port 0/1/3
!
dial-peer voice 999010 pots
service mgcpapp
port 0/1/0
!
dial-peer voice 6 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 17772253754
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 7 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 1845205554[4-5]
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
!
!
sip-ua
credentials username 17772253754 password 7 106C1B49111F17194D realm callcentric.com
authentication username 17772253754 password 7 08035E1E1D11000553 realm callcentric.com
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:callcentric.com expires 3600 port 5060 transport udp
registrar dns:callcentric.com expires 3600
sip-server dns:callcentric.com
host-registrar
!
!
!
line con 0
line aux 0
line vty 0 4
password cisco
login
transport input all
!
scheduler allocate 20000 1000
ntp server 199.102.46.72
ntp server 23.227.162.123 prefer
end

exit

whatuusay1 · ‎02-15-2014

Here are the sip profile screenshots.

Thank you,

Andrew

Manish Prasad · ‎02-15-2014

Can you reset SIP trunk once and then test a call.

Thanks

Manish

whatuusay1 · ‎02-15-2014

I reset the trunk and retested (same failure) - I've attached another set of ccsip all debugs (post sip trunk reset).

Thank you,

Andrew

Manish Prasad · ‎02-15-2014

Received:

SIP/2.0 407 Proxy Authentication Required

v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bKD58157A;rport=49552;received=24.123.98.94

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

Can you also remove session target sip-server from dial-peer 6.

dial-peer voice 6 voip

description ## INBOUND CALL from ITSP ##

session protocol sipv2

session target sip-server

Thanks

Manish

Ayodeji Okanlawon · ‎02-16-2014

I have taken some time to look in detail into the logs and here is my summary...

When you configure ooutbound-proxy dns:callcentric.com, all outbound SIP request is sent to that domain (call centric.com)

you need to remove the command "outbound-proxy dns:callcentric.com"

the summary below explain why

++receive INVITE from ITSP+++

Received:

INVITE sip:17772253754@192.168.1.203:5060 SIP/2.0

v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-11f85db5478eab1a1d6560665917bdc9

f: <18165297500>;tag=3601520592-546043

t: <>18452055544@ss.callcentric.com>

i: 3308051-3601520592-546012@msw2.telengy.net

+++Next CUBE sends a trying to ITSP..Before it does that it needs to establish a TCP connection+++

NB: CUBE has sent the call back to ITSP on port 5080, because thats the port used for the incoming connection+++

008382: Feb 16 00:23:12.949: //3342/A5A3315A8319/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x49E83E84 to default port=5080

008383: Feb 16 00:23:12.949: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:204.11.192.159, rport:5080 with laddr:192.168.1.203

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-11f85db5478eab1a1d6560665917bdc9

From: <18165297500>;tag=3601520592-546043

To: <>18452055544@ss.callcentric.com>

+++Next CUBE needs to send an INVITE out to CUCM+++

NB: The target host and the outbound Host are different...The target host is who the call is for, the outbound host is who the call should be sent through+++

08432: Feb 16 00:23:12.965: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : 192.168.1.200 target_port : 5060

008433: Feb 16 00:23:12.965: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: outbound_host : callcentric.com outbound_port : 5060

++++Because your outbound_host is a dns entry, cube needs to resolve the DNS for the domain+++

008547: Feb 16 00:23:12.981: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_DNS_RESOLVE

008548: Feb 16 00:23:12.981: //3343/A5A3315A8319/SIP/State/sipSPIChangeState: 0x49630448 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_SENT_DNS)

+++CUBE does a SRV DNS query (looks like your DNS is configured for SRV++++++++++++

008554: Feb 16 00:23:12.985: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: TYPE SRV query for _sip._udp.callcentric.com and type:1

++++Next DNS server returns all the SRV records for _sip._udp.callcentric.com++++++++

008555: Feb 16 00:23:13.057: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha18.callcentric.com

008556: Feb 16 00:23:13.057: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008557: Feb 16 00:23:13.057: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha19.callcentric.com

008558: Feb 16 00:23:13.057: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008559: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha11.callcentric.com

008560: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008561: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha12.callcentric.com

008562: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008563: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha15.callcentric.com

008564: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008565: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha16.callcentric.com

008566: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008567: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Server Name alpha17.callcentric.com

008568: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Priority 20 Weight 0 Port 5080

008569: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: Selected Server is alpha18.callcentric.com

008570: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_query: TYPE A query successful for alpha18.callcentric.com

008571: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_a_query: ttl for A records = 263 seconds

+++Next DNS sends the ip address for the chosen device..204.11.192.170 (alpha18.callcentric.com)++

008572: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: IP Address of alpha18.callcentric.com is:

008573: Feb 16 00:23:13.061: //-1/xxxxxxxxxxxx/SIP/Info/sip_dns_type_srv_query: 204.11.192.170

+++NEXT CUBE sends a UDP connection to 204.11.192.170++++++++++

008620: Feb 16 00:23:13.077: //3343/A5A3315A8319/SIP/Transport/sipSPISendInvite: Sending Invite to the transport layer

008622: Feb 16 00:23:13.077: //3343/A5A3315A8319/SIP/Transport/sipSPITransportSendMessage: msg=0x4A9AEC54, addr=204.11.192.170, port=5080, sentBy_port=0, local_addr=192.168.1.203

008626: Feb 16 00:23:13.077: //-1/xxxxxxxxxxxx/SIP/Transport/sipCreateConnHolder: Created new holder=0x4A9B0760, addr=204.11.192.170; nailed=FALSE

008627: Feb 16 00:23:13.077: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostRequestConnection: Posting UDP conn create request for addr=204.11.192.170, port=5080, context=0x4840BD14

+++CUBE succesfully created a connection with 204.11.192.170+++++++++++

008638: Feb 16 00:23:13.081: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessConnCreated: connection instance created for addr:204.11.192.170, port:5080 local_addr=192.168.1.203 local_port=49552

Next CUBE procceeds to send an outbound INVITE

008646: Feb 16 00:23:13.081: //3343/A5A3315A8319/SIP/State/sipSPIChangeState: 0x49630448 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_SENT_INVITE, SUBSTATE_NONE)

++++++++++++but the INVITE appears to go to cucm when infact it wasnt sent there..it was sent to ITSP!!!!!!!!!!

Sent:

INVITE sip:17772253754@192.168.1.200:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bKD58157A

From: <>18165297500@callcentric.com>;tag=B46262C-230A

To: <17772253754>

++++NEXT we get a response back from your ITSP requesting authentication++++++++

008670: Feb 16 00:23:13.181: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog

008671: Feb 16 00:23:13.181: //3343/A5A3315A8319/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 407 Proxy Authentication Required

v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bKD58157A;rport=49552;received=24.123.98.94

f: <>18165297500@callcentric.com>;tag=B46262C-230A

t: <17772253754>

i: A5AF6643-960911E3-831FFFD7-951695DD@callcentric.com

CSeq: 101 INVITE

Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="b69582701157bab55830417b7b62def9", opaque="", stale=TRUE, algorithm=MD5

l: 0

So, even though it looks like the call was sent to cucm, it was never sent to CUCM, it was sent to ITSP for onward forwarding to CUCM!!!!!!!!!!!!!!!!

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whatuusay1 · ‎02-16-2014

Aok - Hugely helpful notes!! you were right - I removed the "outbound-proxy dns:callcentric.com" command and calls now ring through to my CUCM registered SCCP phone. When I answer the CUCM phone I receive a couple seconds of silence and then reorder tones. The calling party (18165297500) receives ringback and then an error that the party isnt avaialble from the local telco.

I've attached ccsip all debugs and cucm traces for the failed call as well as a current running config from the cube gateway.

Thank you so much for the assistance thus far - we're making progress

Calling number is 18165297500

Called number is 8452055544 (although the callcentric username 17772253754 is what you'll see in the logs)

Time of call 1:24PM CST

SCCP Phone Mac 001EBE91435C

Thanks,

Andrew

Ayodeji Okanlawon · ‎02-16-2014

Excellent..We are making proigress...lookslike you dont like me though You havent rated any of my posts! My last post took all of my brains to figure out ! lol

Okay, Now the call progressed as expected and CUBE sends a 200 OK back to your ITSP..but they didnt respond to the 200 OK. You need to get on the phone with them and find out why they arent sending an ACK to your 200 OK. Come back and update us once thats done. We will then work on the sip profile to sort out the how to copy the TO header to the Request URI

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

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whatuusay1 · ‎02-16-2014

Aok - My appologies, I think i didnt understand how the rating was working (I was rating the wrong posts). I rated your responses - very very helpful :-) Thank you!

I sent a note to the ITSP (Callcentric) and will respond once I hear back from them. They do not have a CUCM/CUBE template but they have been helpful in troubleshooting. They too saw the reinvite back (from the proxy statement) but weren't sure how to explain it.

It makes me happy it was a tough problem - I hate to ask for help on th easy ones

Thanks again and will reply shortly.

Andrew

Manish Prasad · ‎02-16-2014

Deji

My last post took all of my brains to figure out - +5 for this.

whatuusay1 · ‎02-17-2014

AOK - I received the following back from the ITSP. It looks like they dont like the require timer statement.

From the trace provide we do see the following:

Require: timer

Can you please modify your configuration so that you are no longer requiring session timers? Once you have done so retest calling then update this ticket.

If you have any other questions, please feel free to ask. Thank you.

Thank you,

Andrew

Ayodeji Okanlawon · ‎02-17-2014

Ok..Try this...and test again. (if it doesnt work please send debug ccsip messages only)

voice class sip-profiles 5

response 200 sip-header Require REMOVE

voice service voip

sip

sip-profiles 5

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

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whatuusay1 · ‎02-17-2014

Aok - We have sucess! Inbound calls are now sucessful! Talk about a happy day..I still have the DID routing issue, and i need to get outbound calls to include username or a DID on my account to sucessfully route .. but I dont think ive ever been so happy to hear a test call go through.

Here's the info they provide for outbound Caller ID.

To send the caller ID number of a DID on your account or an already verified number within the SIP INVITE message of an outbound call you will need to have your user agent (UA) attach any of the following headers from highest priority to lowest priority, which are supported by most IP PBX's (check your vendor's documentation for support), to your outbound calls:

REMOTE-PARTY-ID
P-ASSERTED-IDENTITY
P-PREFERRED-IDENTITY

Thank you,

Andrew

Ayodeji Okanlawon · ‎02-17-2014

For the inbound call..try this...and test to see that calls are sent using the TO header..

voice class sip-profiles 5

request INVITE sip-header TO copy "<>" u01

request INVITE sip-header SIP-Req-URI modify “.*@(.*)” “INVITE sip:\u01@192.168.1.200:5060 SIP/2.0

For outbound calls, I need to know the range of your DDI...What should the extensions map to or do you just want to present a single DDI to your provider for all calls? If that is correct, then what is the DDI you want to use

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whatuusay1 · ‎02-18-2014

Aok - Thanks, I'll give that a shot and see if the calls route correctly (I wont be able to test until this evening).

I have the following DID's

18452055544

18452055545

Outbound I'd want to see where the call was coming from and correctly insert that into the header - I assume something like

sip profile 6

request INVITE sip-header From copy "<>" u01

request INVITE sip-header P-Asserted-Identity modify “.*@(.*)” “INVITE sip:\u01@192.168.1.200:5060 SIP/2.0

And then put that on the outbound dial-peer to Callcentric (ITSP)?

Thanks again for all the assistance

Ayodeji Okanlawon · ‎02-18-2014

voice class sip-profile 6

request INVITE sip-header P-Asserted-Identity modify “(.*)@(.*)” "18452055544@\2"

request INVITE sip-header Remote-Party-ID modify “(.*)@(.*)” "18452055544@\2"

Then apply to dial-peer towards ITSP or apply it to existing sip profile 5

sip profiles 5

request INVITE sip-header P-Asserted-Identity modify “(.*)@(.*)” "18452055544@\2"

request INVITE sip-header Remote-Party-ID modify “(.*)@(.*)” "18452055544@\2"

Please rate all useful posts

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

Please rate all useful posts