Cisco 7821-K9 2 Lines Asterisk

[ak2018]

After few days of messing around with the configuration I finally managed to connect cp-7821 to asterisk pbx, however, I can't seem to get it to connect to 2 different sip providers, since the 7821 has 2 lines technically this should be possible, but since this is a workaround for the enterprise phone would it still be possible to use 2 lines?

[Leo Laohoo]

Are you trying to connect the phone to 2 SIP accounts or is there an Asterisk box between the phone and the SIP accounts?

[ak2018]

I'm trying to connect to 2 different sip accounts.

[Leo Laohoo]

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@ak2018 wrote:

I'm trying to connect to 2 different sip accounts.

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Is there an Asterisk server between the phone and the two accounts? 

What I am saying is the Asterisk server is already connected to the two accounts and the phone is attempting to connect to the same two accounts.  

If this is the case, it will depend entirely on the service provider.  A lot of voice providers will only allow one authentication.

[ak2018]

2 different sip providers, so there will be 2 different sip addresses.

[balaji.bandi]

To understand better you have Phone with 2 Lines, that Phone register with your Asterisk Server.

Asterisk Server have 2 Sip providers, and from phone you are not able to make calls ?

is this correct ? is so then go to next level, if this wrong my understanding please correct here.

1. if so please check is the Asterisk register with Sip provider. if they are register

2. you have correct dial plan in Place to route the calls ( if possible share the dial plan)

3. check the asterisk cli log when you try to make calls what is the error you getting.

[ak2018]

2 different sip providers that means 2 different sip server addresses, not 2 different extension coming from the same sip server. For example, line 1 is connected directly to Asterisk PBX, so that would be one server address, and line 2 would connect directly to sipgate and that would be my second sip server address.

Line 1 = 192.168.3.100

Line 2 = sipgate.co.uk

[balaji.bandi]

since the 7821 has 2 lines technically this should be possible, but since this is a workaround for the enterprise phone would it still be possible to use 2 lines?

Coming back your question, are you setup to register with 2 SIP Servers Asterisk and Sipgate.

As long as your dial plan to make it different, you should be good. ( make sure Phone has support with 2 SIP Servers)

[ak2018]

I don't think you fully understand, the workaround is to put the sip server address at the top 

<callManagerGroup>
   <members>
    <member priority="0">
     <callManager>
      <ports>
        <ethernetPhonePort>2000</ethernetPhonePort>
        <sipPort>5060</sipPort>
        <securedSipPort>5061</securedSipPort>
      </ports>
  <processNodeName>[SIP SERVER ADDRESS]</processNodeName>
</callManager>

You can only put 1 server address. Is there a way to put second server address?

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[balaji.bandi]

I don't believe so, you have an capabilities to register with 2 SIP Servers with this Phone.

But may be work around, you can register 2 lines with Asterisk, and when the 2nd line call come on you can redirect the call from Asterisk to Sipgate

make sense ?

[ak2018]

I'm aware of redirecting sipgate via asterik, but I wanted a connection direct to sipgate and bypassing asterisk. I can see now that it is not possible.

[balaji.bandi]

You can setup Asterisk as proxy so media will be directed to Sipgate, once authentication done.

[Leo Laohoo]

<processNodeName>[ASTERISK]</processNodeName>
<processNodeName>[SIP GATE]</processNodeName>

What happens if you put two?

I remembered reading something like this and the phones can support up to 4 IP addresses.

[ak2018]

It uses call manager for the sip server address

<proxy>USECALLMANAGER</proxy>

<callManager>

        <ports>

        <ethernetPhonePort>2000</ethernetPhonePort>

        <sipPort>5060</sipPort>

        <securedSipPort>5061</securedSipPort>

                  </ports>

        <processNodeName>[SIP SERVER ADDRESS]</processNodeName>

               </callManager>

So line 1 calls call manager to provide the proxy address, line 2 can't call the same call manager because it uses a sip server I don't want to use for line 2.  

[Leo Laohoo]

        <callManagerGroup>
            <tftpDefault>true</tftpDefault>
            <members>
                <member priority="0">
                    <callManager>
                        <ports>
                            <ethernetPhonePort>2000</ethernetPhonePort>
                            <sipPort>5060</sipPort>
                            <securedSipPort>5062</securedSipPort>
                        </ports>
                        <processNodeName>ASTERISK</processNodeName>
                    </callManager>
                </member>
				<member priority="1">
                    <callManager>
                        <ports>
                            <ethernetPhonePort>2000</ethernetPhonePort>
                            <sipPort>5060</sipPort>
                            <securedSipPort>5062</securedSipPort>
                        </ports>
                        <processNodeName>SIP GATE</processNodeName>
                    </callManager>
                </member>
            </members>
        </callManagerGroup>

How about this?