- asterisk server with 4xPRI E1 interface card (TE405P)
- Telco PRI trunk connected to port 1
- Alcatel 4200 E PBX connected to port 2
- 3 Sip phones defined as Sip extensions.
- PBXinaflash, including Freepbx 126.96.36.199. and Centos 5.1
- soon to be added: Sip trunks
Eventually the asterisk server should ideally function a.o. as a sort of connection point between my existing ISDN-30 PBX/Telco and the new Sip Voip phones and later Sip trunks.
Connected to the PBX are telephones which are selected by DID/DDI number, not by channel number. They pool the 30 available ISDN channels in the E1 ISDN30 trunk.
In order to be able to connect to the phones behind the PBX I have defined them as ZAP extensions, with a dial setting like ZAP/g1/xxxxxxxxx, with xxxxxxxxx being the DID number. This works,
However, since I have defined them as ZAP extensions they generate entries in zapata_additional.conf, which creates problems because in real life these extensions have no specific channels assigned to the, while the generated enrtres in zapata_additional.conf expect them to.
I can work around this by commenting out the include of zapata_additional.conf in zapata.conf, and including a zapata-custom.conf where I have defined my trunks manualy instead, but this looks a bit kludgy. Also, I see some messages in my asterisk log like ‘Failed to read gains: invalid argument’ which might be related to this.
Is there a proper way to define ZAP extensions which use pooled channels?
Another related question: where can I define the specific settings for my E1 trunks, like the switchtype (in my case euroIsdn)? I can’t set this in Zap Trunks. Or are these definitions somehow expected to be generated by genzaptelconf in zapata-auto.conf? In my case they weren’'t so I had to define them in zapata-custom.conf