SIP TRUNK without CUBE

[csco11030279]

hi;

is there anyone configured cisco communication manager for direct calls to SIP Trunk provided by the Service Provider (Without CUBE)?

I am looking for configuration steps

Thanks

[barry]

Hi

It all depends on how you want to configure the SIP trunk. CUCM only provides VERY basic support for SIP trunking - specifically it does not support registration / proxy.

I have been able to get a trunk between CUCM and the ITSP working, but only when using IP address authentication. Note that CUCM's validation of IP addresses (without using SRV) is very limited - in that you have to specify each explicit IP address your ITSP can attempt to signal you on - you can't set up wildcard masks.

IMHO, it's much better with a CUBE.

HTH.

Barry Hesk
Intrinsic Network Solutions

[csco11030279]

thank you for the answer. to have CUBE will be costy and and will take long time .

so what do you think that I need to list down from the SP to check the comatability . in fact this is my first time I do such cenario.

[barry]

Hi

Key things that I talk to ITSPs about with SIP trunking are:

1. How they perform authentication. If SIP Registration or Proxy are required then you can't use CUCM directly. CUCM does support SIP digest authentication as well as IP address authentication, although as per my previous post CUCM can only authorise incoming connections by specifying each individual IP address that the ITSP may use (if DNS SRV isn't in place).

2. Other things that come up are codecs (G711/G.729) and DTMF. Sometimes Early Offer is required.

Other than that, I've never really had too many issues getting it to work. Authentication of the trunk is the painful piece.

HTH.

Barry Hesk
Intrinsic Network Solutions

[Leo Laohoo]

Do you want a CUCM between your phone and the voice service provider?

I mean do you specifically want CUCM or are you looking for an alternative?

[csco11030279]

yes I want it to be there ,

the most imortant thing now that I dont want to use gateway or cube ,

[kaja_2kj3]

Hi

    Also it is not secure, to expose your cucm.

thanks

[csco11030279]

no problem . but how do you have configuration steps

[barry]

Basically

1. Create a SIP trunk security profile, and configure things such as transport type (TCP/UDP) for incoming / outgoing connections. Also configure digest authentication if your ITSP requires it.

2. Edit the default SIP profile. 99.9% of this should be ok, but you may need to adjust some timers if your ITSP requires it.

3. Create a SIP trunk on CUCM pointed at the IP address(es) of the ITSP SIP endpoint(s). Configure DTMF / Codec accordingly. Ensure the SIP trunk(s) have a CSS that can access internal numbers.

4. Create any additional SIP trunks for each of the IP address(es) that your ITSP may signal from. There is a limit to how many destination addresses you can configure on a single trunk.

5. Create translation patterns for incoming calls as required.

6. Create Route Patterns for outbound calls as required pointing to the trunks.

As an aside, I would always place a firewall in between CUCM and an ITSP. If the firewall needs to NAT CUCM's IP address (e.g. it is an Internet attached ITSP) you must have SIP NAT ALG support on the firewall.

Test and see how it works. Packet capture on CUCM via CLI is really useful to see session flows.

Barry Hesk
Intrinsic Network Solutions

[csco11030279]

Hi

this is what I get from the SIP SP

any idea ???

Basic SIP trunk- PBX setup RTTS TT # PR00003467354

Main Service information

We provide the customer with the information below:

• •1.       Customer IP  : 172.29.29.142 /30 and the VLAN ID : 2622
• •2.     STC gateway IP : 172.29.29.141/30 
• •3.     SIP server IP :   10.209.4.58

Customer range or numbers: XXXXXXXXX-YYYYYYYY

Connectivity trouble shooting.

• •1.     Be sure that your link is up and the customer IP is defined in the PBX side**.
• •2.     Be sure that you can reach the STC gateway.
• •3.     Be sure to define the Sip server route via the gateway by defining the route subnet address with subnet mask of  10.209.4.56

255.255.255.252.

• •4.     Be sure that you can reach the SIP server.
• •5.     Be sure to activate the heart beat*** functionality in your system so that our SIP server register your PBX as active.

** in some cases you need to setup the VLAN configuration

*** in some PBXs it is under the name SIP OPTIONS PING

Standard Parameters

You have to be sure that the following parameters are defined correctly to make a call:

Protocol= SIP

SIP Port = 5060

Transport Protocol=UDP

Voice Codec= G711 A-Law

DTMF = IN-Band DTMF with RFC2833

Call troubleshooting

• •1-    Be sure that your range is installed correctly in the two formats
  • •a.     7 digits
  • •b.     9 digits
• •2-    the SIP server authenticities your call based on the number and the customer IP. So be sure that you are sending your INVITE message holding the correct source number and source IP.
• •3-    Be sure that you are sending packets toward the correct destination IP of SIP server.
• •4-    Be sure that the sequence of the SIP packets in any session is correct.

• •5-    Be sure that you are using the correct Voice codec of G711 A-Law.
• •6-    Any difficulties after making the call correctly would be happened due to one of the following:
  • •a.     Crackling or hard to hear the second party: insure of the internal LAN Qos.
  • •b.     One way voice: could be happened due to customer’s internal routing issue. You need to be sure that the default route is the STC gateway.
  • •c.      Can’t call internationally: check the INVITE destination number length or PBX authority of making the International calls.
  • •d.     DTMF problems: check the DTMF RFC correctly, be sure that your PBX is sending the correct telephony event packet toward the STC.

Fax configuration

• •Ø Use T38 codec for FAX. (Should be G711A pass through).
• •Ø EC cancellation setting (default settings are OK).
• •Ø Type of Fax used (Customer should be aware of type of fax High Speed or low speed).

Encryption configuration

• •Ø the encryption box should be compatible with fax  machine, 
• •Ø in NETWORK prospective normal fax and encrypted fax are not different.

Converters configuration

• •Ø There might be many type of converters serving to the customers, most commonly used are TDMOIP (TDM over IP) used to convert Traditional signaling (R2, Qsig,) to IP
• •Ø We strongly advice to let the vendor support in installing and interconnecting the converters.

Further troubleshooting

• You can contact our 24/7 STC SIP helpdesk at 01-4432549 on Email: ENOC_Fixed@stc.com.sa.
• You need to mention the name of your circuit, your range numbers and your IP address.

[ibtihaj2010]

Hi Barry,can u pls help me with sum screenshots for the Sip to Itsp authentication?

also some screenshots for translations for incoming calls

thnx in advance

i am about to directly do this scenario to make a sip directly to Itsp without a cube router

[joshua.gertig]

We've just implemented SIP trunks with no CUBE. We used Level3, with IP VPN to CUCM 8.6.2. Let me know if you'd like me to post sanitized screen shots of the CUCM set up.

[Leo Laohoo]

Other than Cisco 7960G, 7970G, I've got an Asterisk running on a Raspberry Pi.  No complaints. 

[fmosley823]

Hi Joshua, 

I know this is an old post, but if you still have this setup, would you be able post?  I'm interested in connecting cucm directly via sip trunk to my ITSP (voip.ms).  

Thanks

Frank

[ibtihaj2010]

hi Joshua

can u help me with sum screenshots for the sip registration to Itsp without cube router pls

i m about to deploy this solution n need help , thnx

also screenshots of translation of incoming calls can help