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Hi Guys, I would like to know if BCM 5.0 supports receiving and processing the SIP REFER method and also the INVITEs with Replaces header inside. Thanks.

Hi Guys, We have the following topology: SIP PRXY ---- CS1K(1)---- NRS ----- CS1K(2) The SIP proxy we use isn't able to communicate with the Nortel NRS (redirect mode), so we use the CS1K(1) as tandem PBX -- all the SIP signaling between the SIP PRXY and the CS1K(2) goes trough the CS1K(1) --...

Hi guys, Do somebody know how to recover/reset a NRS password ? Thanks.

So, let me see if I understood, for instance for a simple call, user A at Asterisk call user B at CS1000, we will have the INVITE from A to B, then 180 and 200 OK from B to A and finally the ACK from A to B. The described signaling will consume 2 SIP trunk license ? Regards.-

Hi Guys, I have a CS1000 connected to an Asterisk box by a SIP Route with x trunks inside. The questions are: 1) How many (x) SIP trunk licenses I need to purchase in order to establish 10 simultaneous SIP calls between the CS1000 and the Asterisk box? 2) Is it needed one SIP trunk license...

Te number received is [002]: 01110000 Information Element: Called Party Number: 2 IE Length : 4 octets 3 1------- Extension bit : Not Continued -100---- Type of number : Subscriber number ----0001 Numbering plan : ISDN/telephony numbering - Rec. E.164...

The number of digits to absorb is set to 0, also the outgoing calls are working.

Hi, A have configured a E1 PRI ISDN to connect to the PSTN, I can make outgoing calls, but the incoming calls are not working. I performed the following command to get traces edt trace tsp l2l3 3 1 2 1 and the traces shows the following error, -0101100 Cause code: requested cct/chan unavail...

I am using the 5212 just to get a SIP phone working, we plan to have 3rd party sip phones in our deployment.

Hi guys, I am trying to register a 5212(SIP firmware) with a 3300 cx, the phone sends the REGISTER but the PBX responds with a ICMP (port unreachable, i have checked and the port is configured ok (5060 UDP). When a list the licences availables i get 2 for SIP phones but i get nono for IP...

Actually the BCM is the one which generates the SIP messages with the unique--boundary-1 info, the guy which don't support that feature is the 3rd party SIP proxy.

Hi guys, Do you know how to eliminate the unique--boundary-1 info from the SIP messages in the BCM400. I am trying to connect the BCM to a SIP proxy which do not support the -unique--boundary-1 info. Thanks.

Hi, I am trying to establish a trunk between a SIP proxy and a BCM 400 (v 4.0) using TCP, I have configured the BCM 400 and also the SIP proxy. When the proxy sends the SYNC to establish the TCP connection the BCM rejects the handshake by sending a RST. Do you know if it could be a...

Hi Guys, I have a CS1k configured to use PCA, to twin a deskset with a UA managed by a SIP proxy, when I ring the deskset the CS1K generates the INVITE to the UA bahind the proxy with a extra CRLF at the end of the SDP, due that the UA reply with a 400 Bad request. Do you know if the SDP RFC...

Hi guys, I would like to know how to setup the CS1000 in order to have the media traffic between tow phones going trough the PBX (known as media shuffle or direct ip-ip. Thanks.

Hi all, When I send a basic call request from our SIP server through the CS1K, targeting a PSTN destination, the SIP INVITE is responded to with two 183 Session Progress Provisional response messages from the CS1k. The interesting thing is that the first 183 response contains an SDP message...

Hi, Do you know if is it possible to connect three CS1000 without use NRS, but using IP trunks. Thanks.

How does it work, I have Element Manager and i see two prompts: Incoming digits: (0 - 9999999) Conceverted Digits: (0- 9999999) I want to convert 3515705xxx to 5xxx, wath i should enter in each prompt ? (x = any digit 0-9) Thanks

Hi all, I have a CS1k connected through a SIP trunk to a SIP Gateway which handles 10 digits extensions, the CS1k has 4 digits extensions, so I want to delete the first 6 digits of the calls which come from the SIP Gateway (I am not using the NRS). Do you know some way to delete the digits...

Hi All, Do you know how to implement simultanues ringing without use PCA nor Mobile Extension (UEXT), the scenario is as follow: SIP Trk Ext 0 ----- CS1000 (1) ---------- PBX (2) ----- Ext 1 The Ext1 is configures as a DSC (Distant Stering Code on the PBX 1) I want...