Hello!
I have an IPLDK-20 I want to use with a SIP trunk.
Background:
Have used the device with 2 ISDN numbers until now, but want to switch to SIP. I have ordered a SIP trunk from a provider and have 3 numbers available and ready to use.
I have about ten LPD-7024D telephone units, and to call out they all had to push any of two buttons to get a “line out” (none of the phones had any direct numbers). I would like the same functionality with the SIP trunk. I don’t have any “real” SIP phones, and don’t have any plan to add any.
I don’t have any good understanding on how the whole thing is supposed to be set up, and haven’t found any professional service in my area I can hire to help me with the setup. The SIP trunk provider only supports “their end”, and not specific hardware equipment at the end users locations.
Current situation:
I have read several setup guides, but am still uncertain how to configure it (since I still don’t understand the overall way of thinking). I have updated the IPLDK-20 firmware to C.9Ec.
When I look in the configuration I see various slots:
1. Hybrid
2.
3.
4. DTIB8
5. PRIB
6. LCOB4
7. VMIBE
8.
9. VOIB
10.
Settings I’ve made (without beeing sure if they’re right)
PGM 140: Set CO 1-16 (all) to: ISDN DID
PGM 322: Set CO 1-16 (all) to: group 1, type PSTN, Gatekeeper OFF, VOIP mode SIP, DTMF Inband
PGM 340/341: ip,gateway/subnet/dns to the corresponding network settings, firewall to same as gateway, VOIB mode SIP.
SIP attr 1: Proxy server: my trunk providers sip server. Use outbound proxy NOT ticked(?). Domain: my trunk providers sip server.
Open questions:
- If I look in CO Line list, I see 16 lines. Are those software configured, or bound to specific hardware ports? If they’re hardware bound, how do I know which ports are bound to the VOIB module?
- Where do I configure the phone numbers to use in the SIP trunk?
- If I press a button on a phone to open a line like I did earlier with the ISDN lines, I now instantly get an error tone. In the phones display I see (translated from swedish) “R-BUTTON” on the 3rd row in the display, then after maybe 0.5s I see “No access” on the 1st row. Is this an indication that this station doesn’t have access to something, or is it that the SIP trunk configuration isn’t yet properly setup and working?
Further questions (for the future if I get the basic operation going):
With the previous ISDN lines we had some automatic queue enabled. Lets say our main phone number is 1111 and that line is busy and another call is made to it, the call would automatically get redirected to another available line (=as long as we had an available line, the caller wouldn’t get a busy tone). Is this a configuration made in the IPLDK-20, or was that a feature our previous provider added?
Thanks for any help!
I have an IPLDK-20 I want to use with a SIP trunk.
Background:
Have used the device with 2 ISDN numbers until now, but want to switch to SIP. I have ordered a SIP trunk from a provider and have 3 numbers available and ready to use.
I have about ten LPD-7024D telephone units, and to call out they all had to push any of two buttons to get a “line out” (none of the phones had any direct numbers). I would like the same functionality with the SIP trunk. I don’t have any “real” SIP phones, and don’t have any plan to add any.
I don’t have any good understanding on how the whole thing is supposed to be set up, and haven’t found any professional service in my area I can hire to help me with the setup. The SIP trunk provider only supports “their end”, and not specific hardware equipment at the end users locations.
Current situation:
I have read several setup guides, but am still uncertain how to configure it (since I still don’t understand the overall way of thinking). I have updated the IPLDK-20 firmware to C.9Ec.
When I look in the configuration I see various slots:
1. Hybrid
2.
3.
4. DTIB8
5. PRIB
6. LCOB4
7. VMIBE
8.
9. VOIB
10.
Settings I’ve made (without beeing sure if they’re right)
PGM 140: Set CO 1-16 (all) to: ISDN DID
PGM 322: Set CO 1-16 (all) to: group 1, type PSTN, Gatekeeper OFF, VOIP mode SIP, DTMF Inband
PGM 340/341: ip,gateway/subnet/dns to the corresponding network settings, firewall to same as gateway, VOIB mode SIP.
SIP attr 1: Proxy server: my trunk providers sip server. Use outbound proxy NOT ticked(?). Domain: my trunk providers sip server.
Open questions:
- If I look in CO Line list, I see 16 lines. Are those software configured, or bound to specific hardware ports? If they’re hardware bound, how do I know which ports are bound to the VOIB module?
- Where do I configure the phone numbers to use in the SIP trunk?
- If I press a button on a phone to open a line like I did earlier with the ISDN lines, I now instantly get an error tone. In the phones display I see (translated from swedish) “R-BUTTON” on the 3rd row in the display, then after maybe 0.5s I see “No access” on the 1st row. Is this an indication that this station doesn’t have access to something, or is it that the SIP trunk configuration isn’t yet properly setup and working?
Further questions (for the future if I get the basic operation going):
With the previous ISDN lines we had some automatic queue enabled. Lets say our main phone number is 1111 and that line is busy and another call is made to it, the call would automatically get redirected to another available line (=as long as we had an available line, the caller wouldn’t get a busy tone). Is this a configuration made in the IPLDK-20, or was that a feature our previous provider added?
Thanks for any help!