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SIP Provider Registration

MohamedHussen
Level 1
Level 1

Hi Team

 

I have 7960 phone, converted to SIP and want to register directly to SIP provider, 

My set up is Phone>>3560switch>>> 2851>>> ADSL >>>WWW

 

Device ID: SIP0013C4AACF8B <<<<<<<<<<<<<<<<<<< Phone details - sh cdp neigh
Entry address(es):
IP address: 192.168.5.14
Platform: Cisco IP Phone 7960, Capabilities: Host
Interface: FastEthernet0/3, Port ID (outgoing port): Port 1
Holdtime : 136 sec

Version :
P0S3-07-5-00 <<<<<<<<<<<<<<<<<<<<<<<<<< SIP Version on the phone

advertisement version: 2
Duplex: full
Power drawn: 6.300 Watts
Management address(es):


============================================
Current configuration : 171 bytes
!
interface FastEthernet0/3 <<<<<<<<<<<<<<<<<<<<< Phone connected Port on the switch
description PBX VLAN
switchport access vlan 30
switchport mode access
switchport voice vlan 5
random-detect
spanning-tree portfast
end
=========================================

SIP Config File
SIP Configuration Generic File (start) <<<<<<<<<<<<<<, phone SIP file

<deviceProtocol>SIP</deviceProtocol>
<sshUserId>xxxx</sshUserId>
<sshPassword>xxxx</sshPassword>
<loadInformation>P0S3-07-5-00</loadInformation>
<devicePool>

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

XMLDefault.cnf.xml <<<<<<<<<<<<< default xml file

image_version: "P0S3-07-5-00"

;Sip default configuration file
#Image Version image_version:P0S3-07-5-00 ;
#Proxy address
proxy1_address: x.www.sip.com ;
#Default Codec
preferred_codec :g711ulaw ;
#Enable Registration
proxy_register :1 ;
#Registration expiration
timer_register_expires :3600 ;

 

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

Errors

*Nov 25 23:14:12.998: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
*Nov 25 23:14:12.998: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIFindRegisterCcb: *****CCB NOT found in UAS Request table. ccb=0x00000000
*Nov 25 23:14:12.998: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x48960934) with key=[16015] to table
*Nov 25 23:14:12.998: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 192.168.30.12,Port 64304, Transport 1, SentBy Port 64304
*Nov 25 23:14:12.998: //-1/1336970CBF1E/SIP/State/sipSPIChangeState: 0x48960934 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
*Nov 25 23:14:12.998: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 192.168.30.12,Port 64304, Transport 1, SentBy Port 5060
*Nov 25 23:14:12.998: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT
*Nov 25 23:14:12.998: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 192.168.30.12,Port 64304, Transport 1, SentBy Port 64304
*Nov 25 23:14:12.998: //-1/1336970CBF1E/SIP/Transport/sipSPISendResponse: Sending INFO Response to the transport layer
*Nov 25 23:14:12.998: //-1/1336970CBF1E/SIP/Transport/sipSPITransportSendMessage: msg=0x489CE3D4, addr=192.168.30.12, port=64304, sentBy_port=64304, is_req=0, transport=1, switch=0, callBack=0x00000000
*Nov 25 23:14:12.998: //-1/1336970CBF1E/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Nov 25 23:14:12.998: //-1/1336970CBF1E/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Nov 25 23:14:12.998: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x489CE3D4, addr=192.168.30.12, port=64304, connId=0 for UDP
*Nov 25 23:14:13.002: //-1/1336970CBF1E/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
*Nov 25 23:14:13.002: //-1/1336970CBF1E/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x48960934 key=ZGRmNTNjYmFiZWIzNzI5NDEzMWZkODBmMGFkNTQ1MDc.2002901c9676
*Nov 25 23:14:13.002: //-1/1336970CBF1E/SIP/Event/ccsip_api_register_ind: CCSIP_REGISTER:: Registrar is not enabled
*Nov 25 23:14:13.002: //-1/1336970CBF1E/SIP/Event/sact_idle_new_message_register:
ccsip_api_register_ind return value : SIP_SERVICE_UNAVAIL

*Nov 25 23:14:13.002: //-1/1336970CBF1E/SIP/Transport/sipSPISendResponse: Sending INFO Response to the transport layer
*Nov 25 23:14:13.002: //-1/1336970CBF1E/SIP/Transport/sipSPITransportSendMessage: msg=0x48953264, addr=192.168.30.12, port=64304, sentBy_port=64304, is_req=0, transport=1, switch=0, callBack=0x41519D40
*Nov 25 23:14:13.002: //-1/1336970CBF1E/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Nov 25 23:14:13.002: //-1/1336970CBF1E/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Nov 25 23:14:13.002: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x48953264, addr=192.168.30.12, port=64304, connId=0 for UDP
*Nov 25 23:14:13.002: //-1/1336970CBF1E/SIP/State/sipSPIChangeState: 0x48960934 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)

 

I just want the phone to directly register with the SIP and the router to pass the information not interfere at all. I have attached the router config as well which is basic, your help is appreciated 

1 Accepted Solution

Accepted Solutions
27 Replies 27

Dennis Mink
VIP Alumni
VIP Alumni

you can run call manager express, but you will need to configure the physical phone, using the phones mac, create a DN and associate this with the physical phone.

Please remember to rate useful posts, by clicking on the stars below.

The phone is registered with the CME but I want to register with the ISP, The phone has all the credentials provided by the provider, this information is on the phone and not on the CME, is there anything else i can do.

 

 

Normally, the voice gateway registers with the ITSP. In your case, the gateway is also the phone system = CUCME. So you would register the phone to CUCME and simulteneously register CUCME to the ITSP. This is the normal way of doing things. Then you have the ability to add more phones on the LAN without changing the CUCME <-> ITSP config.

I understand you and than you for your help, in order to configure my CME a SIP trunk port with my SIP provider what do i need to configure, so far I have configured SUP-UA features but nothing has changed, the SIP provider has given me a sample extension as username, a Password and proxy www address, I believe these are the credentials required for each extension that is going to register with the SIP provider, as for the CME to the SIP provider do I need extra information.

 

Again my set up phone>>>switch3560<<<Router2851>>ADSLModel to www

I am doing NAT overload on WAN interface 

 

Thanks

 

Is this for a lab? Give this a go:

 

[sip registration username] = the username you use for sip registration

[sip pass] = the password for sip registration

[sip domain.com] = the sip realm for registration. If you dont have it, you can try without or put in something dummy and then take a debug to see what the ITSP is sending. 

sip.itsp.com = your ITSP registration url

 

sip-ua
credentials username [sip registration username] password [sip pass] realm [sip domain.com]
registrar 2 dns:sip.itsp.com expires 60 refresh-ratio 20 auth-realm [sip domain.com]

 

Then you can check with: show sip register status

No This is not for lab we actually to move to sip 

 

sip-ua
credentials username 1057300 password 11071F131014031B06323A222B38312603 realm 103.93.xx.xx
authentication username 1057300 password 7 030A5D1D0109295B4C11080314190F0814 realm 103.93.xx.xx
retry invite 2
retry register 10
timers connect 100
registrar ipv4:103.93.xx.xx expires 3600
sip-server ipv4:103.93.xx.xx

 

I have tried with software phones like x lite and it works fine but come to cisco not easy, our phones are the old 79xx cant move out yet.

Pls run the debug, attempt registration and then capture all the SIP packets and post back. 

I have added extra captures please see attached 

Hi, 

Something is up with your captures as it's incomplete. I only see one reply from the ITSP there, however it may have the missing info we need. Try this config and then run the captures again:

 

** just add your SIP registration password. I've added the username and the realm **

 

sip-ua
credentials username 1057300 password [sip pass] realm telecube.com.au
registrar 1 ipv4:103.93.71.35 expires 60 refresh-ratio 20 auth-realm telecube.com.au

Post back soon.

It did not like registra 1 i can only registrar ipv4:103.93.71.35 expires 60

 

sip-ua
credentials username 1057300 password 151C0D1A032C23332A2B2424100C120711 realm telecube.com.au
authentication username 1057300 password 7 030A5D1D0109295B4C11080314190F0814 realm telecube.com.au
registrar ipv4:103.93.71.35 expires 60

 

Also under voice register global we have this

mode cme
source-address 192.168.5.1 port 5060
max-dn 5
max-pool 5
load 7960-7940 P0S3-07-5-00
authenticate register
authenticate realm telecube.com.au
tftp-path flash:
create profile sync 001043588101661A
!
voice register dn 1
number 1057300
name STS-SIP
!
voice register pool 1
id mac 0013.C4AA.CF8B
type 7960
number 1 dn 1
username 1057300 password nfvgfhwbxqfckddp
description Mo
codec g711ulaw
no vad

 

Thanks

Ok seems there a few things going on here and confusing the result. 

 

1. Firstly, by default your "ephone-dn" will all try to register against the SIP registrar. To fix this, go into each ephone-dn and add "no-reg primary" after the ephone "number xxxx". This will stop your list of internal extensions trying to register against the ITSP and failing. 

 

2. You don't really need "voice register dn 1". To make things simpler you can remove this until you have registration working and put it back later if you feel you need it. 

 

3. Dont really need this either"

voice register pool 1
id mac 0013.C4AA.CF8B
type 7960
number 1 dn 1
username 1057300 password xxxx

 

4. For the sip-ua config, I have a 2900 so the config is slightly different. Try this for the 2800:

 

sip-ua
credentials username 1057300 password [sip pass] realm telecube.com.au
authentication username 1057300 password [sip pass] realm telecube.com.au registrar ipv4:103.93.71.35 expires 60
sip-server ipv4:103.93.71.35
retry invite 2
retry register 2

 

Then, run the "debug ccsip messages" and make sure you're actually getting responses back. I don't see any replies at all in any of your captures. What firewall do you have? Can you turn on SIP ALG because it's needed to rewrite the 192.168.5.1 IP address into the WAN IP within the SIP packet. The ITSP might be trying to speak to 192.168.5.1 and this might be why you;re not seeing any reply traffic because ALG is off. 

Hi, thanks for your help man, this is real giving me a grief

- I have turned off all the unnecessary configs

- Enabled ALG both TCP and UDP

- updated SIP-UA

I still dont have registration 

 

I have attached the debug CCSIP

 

sip-ua
credentials username 1057300 password 12170301150D04132833352E303E261717 realm telecube.com.au
authentication username 1057300 password 7 030A5D1D0109295B4C11080314190F0814 realm telecube.com.au
disable-early-media 180
retry invite 2
retry register 2
registrar ipv4:103.93.71.35 expires 60
sip-server ipv4:103.93.71.35

 

#sh sip-ua register status
Line peer expires(sec) registered
================================ ========== ============ ==========
.* 1 6 no
1057300 -1 6 no

 

 

voice register global
mode cme
source-address 192.168.5.1 port 5060
max-dn 5
max-pool 5
load 7960-7940 P0S3-07-5-00
authenticate register
authenticate realm telecube.com.au <<<<<<<<<< IS THIS HERE OK?
tftp-path flash:
create profile sync 0014343105722244

 

Contact: <sip:.*@192.168.1.254:5060> <<<<<<<< this is my WAN interface attached to the ADSL modem, I dont have any firewall attached to this network

 

Thanks

Mo

Are you able to ping an internet host? It looks like your internet is not working.

 

https://www.cisco.com/c/en/us/support/docs/ip/network-address-translation-nat/211269-NAT-in-VoIP.html#anc43

Yes i can ping the host 103.93.71.35 normally and also from the source VOIP int

 

STSPBX-R1#ping 103.93.71.35

Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 103.93.71.35, timeout is 2 seconds:
!!!!!
Success rate is 100 percent (5/5), round-trip min/avg/max = 404/415/456 ms
STSPBX-R1#ping 103.93.71.35 r 30

Type escape sequence to abort.
Sending 30, 100-byte ICMP Echos to 103.93.71.35, timeout is 2 seconds:
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
Success rate is 100 percent (30/30), round-trip min/avg/max = 404/417/576 ms
STSPBX-R1#ping 103.93.71.35 r 30 so
STSPBX-R1#ping 103.93.71.35 r 30 source gig
STSPBX-R1#ping 103.93.71.35 r 30 source gigabitEthernet 0/0.5 <<< VOIP Int

Type escape sequence to abort.
Sending 30, 100-byte ICMP Echos to 103.93.71.35, timeout is 2 seconds:
Packet sent with a source address of 192.168.5.1
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
Success rate is 100 percent (30/30), round-trip min/avg/max = 400/417/500 ms

 

 

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