1000 E SIP QUESTION 1

[letron]

HI All,

On a Site with no copper, installed Quintum Tenor fxs device for faxing, able to call to it remotely and internally but cannot dial out, getting fast busy, setup in NRS as gateway endpoint and as a CDP/DSC, i think it is something simple that was not included in DOCS, any help would be great.

Thanks.

 

[Zeus12345]

Comaptibility was tested for CS1K 4.5 software and SIP Trunks...below is from Nortel Website...check the "Configuraiton Notes" also the information bleow pertains to how the SIP and Quintum was tested. Was verified in January of this year. Note there are a couple of patches listed as well you may want to check you have before hand.

http://www.nortel.com/prd/dpp/product/prodpages/certs/cert1749.html

DEV. PRODUCT NAME:
Quintum Tenor AS Series (2-4 Ports) NORTEL PLATFORM TESTED:
Communication Server 1000 E/S/M
INTERFACE/PROTOCOL: SIP
DEV. PRODUCT RLS LEVEL: SIP Module 2.1.0
NORTEL RELEASE LEVEL: Rls 4.5
S/W Patch Release:
MPLR20885, 21049




Compatible Product configuration and testing notes for Quintum Tenor AS Series (2-4 Ports) with Communication Server 1000 E/S/M


The scope of Nortel CS 1000 compatibility testing included:
- Registration
- Basic call (G.711, G.723, G.729)
- Codec and packet size negotiation
- CLID
- Blind call transfer
- Message waiting indication
- Call waiting
- DTMF
- Alternate NRS registration
- T.38 FAX
- FXS and FXO

-Tenor initiated call forwarding, ad hoc conference, and consultative call transfer are not supported at this time. It is expected these will be addressed in a future test event.

- In order to avoid an asymmetric packet/payload size when using G711, CS 1000 must specify packet size in the SDP. For CS 000 to specify packet size in the SDP, all enabled codecs must be configured to use the same packet size.

The following Tenor Configuration Notes are documented in more detail in the Quintum Tenor Configuration Details document which is downloable from the certificate web page.

- Tenor must not include SDP information in the 183-Session Progress message. Inclusion of SDP information in the 183 must be disabled using either the Tenor CLI or GUI interface.
- Tenor must be configured to use the SIP INFO method for DTMF signaling.
- Tenor must be configured to not send multiple 183 messages by turning off PRACK.
- In the absence of DNS server, it may be necessary to statically resolve the Service Domain name used in the NRS. This can be done with a ”DnsStaticHost” entry in the ”var_config.cfg” file in the ”/cfg” directory of the Tenor.
- Failover to secondary or alternate NRS. For successful Tenor failover to the alternate NRS and fallback to the primary, optimal values for ”register Expiry Time” and ”Request Retransmit” were found to be 120 second and 6 retransmit.

 

[sl1tech]

I am having the same problem and am unable to find any useful documentation. Were you able to dial out from the gateway? Can you let me know how you configured NRS to support PSTN dialing?

Thanks,

Dave

 

[I2007]

Have you configured a route entry to the CS1000E in the nrs with the "dial outside" code as cdp or special number...

I2007

 

[sl1tech]

Yes, I have configured the BARS access code 9 as an NRS entry. I have tried it as various E.164, CDP and Private special entires but the calls are still rejected by the gatekeeper.

 

[911guru]

First you need to determine if you are making it past the NRS and the Call Server is failing your call, or the NRS is.

See if you are getting a DCH setup on a call in the Call Server.If you are then you are getting through the NRS. If not you are being blocked at the NRS level.

Now some common mistakes:

INAC on the RDB. Should be there so you don't need the '9' it routes based on TON/NPI of the called number as defined in LD 15 NET_DATA AC2 prompt.

VTRKs are set to NCOS 0 not allowing the call to proceed through NARS/BARS

Number sent by gateway needs to be E.164 NATIONAL and defined properly in the NRS

Is the NRS working for any other function now? DO you have any other sites you can call though it? If not make sure the CS 1000 is properly registered to the NRS and the SIP URI maps are all correct.

911Guru

http://pbx-e911.com

 

[letron]

sl1tech,

I have come a long way since I origanally posted this thread, i now have 16 Quintums working on site and growing, it took alot of screwing around to get this to work, with no help from Quintum or Nortel, if you give me your email address i can forward some info that i think may get you through this.

 

[sl1tech]

No, not making it past NRS. Whenever dialing locally (between call server and Quintum gateway) calls complete successfully both ways and DCH setup/teardown messages are received.

INAC is set to YES on the virtual trunk route.

AC1 is defined for all call types (INTL NPA SPN NXX LOC).

VTRK's are set for NCOS 3, which allows long distance dialing.

I will need to check the Quintum mapping.

No other endpoints are defined in NRS at this time.

 

[sl1tech]

Thanks Letron. My email is dmidcap@mabc.com. I appreciate your help, this has been a frustrating experience.

 

[SneakyHawk]

Hi All,
I am trying to get a Quintum AF working with SIP. I have tested the box with H323 and it works ok, but when I flip it over to SIP things stop working. I can dial to my CS1000E from the Quintum via SIP but cant dial the other way over the SIP. I have set my RLB with entry 0 as the SIP route, and the second entry as H323 but I always get a 404 error off the SIP and the call re-routes to H323. I suspect that the Quintum dial plan is not working correctly but I have been unable to find a mapping for Nortel TON to the Quintum 1xx UPDP. I suspect that 104 may be UDP because the SIP call from the quintum is prefixed with a H but using H323 I do not get the H. If anyone has any information I would be grateful since I am finding information hard to find. Also if Letron could send me the document I would also be very grateful. My email address is charvey@applinet.co.uk

Many thanks

Chris