Around 2013, Swisscom announced their intentions to migrate all residential and all business customers off analogue and ISDN connections on to VoIP. At ENIDAN as well as at home, we have been on ISDN for more than 15 years, but there is probably little doubt that the technology is outdated, we need to progress. In fact, although the connection to Swisscom is ISDN, we switched ISDN off internally back around 2007 and moved everything to VoIP. Moving 100% to VoIP should not be an issue.
To begin with I contacted Swisscom to inquire whether they provided a plain SIP service. “Certainly, we offer an SIP trunk service”. That’s great I thought, but it turned out this was only available with a Swisscom uplink too, which we don’t have and we don’t want. Then I started looking at other VoIP providers. A few years back I had already played with a service from www.sipcall.ch, so I looked them up again.
In June 2015 I decided it was time to get serious, so I created a new account with Sipcall, and routed all our outbound private telephony over that. After a while of testing and trying to get a feel for the Sipcall stability (which is fine), in October 2016 I decided to go the whole way and have our private numbers ported (Vollportierung) to Sipcall. It went very smoothly, no problems at all. Now, about a year later, I have just received confirmation of the porting of our business setup to Sipcall too.
For the business line, we have a couple of issues to resolve –
- sending SMSes
- diverting calls
- sending/receiving faxes.
For critical alerts in the datacentre, we generate SMSes and send them over the Swisscom SMSC (SMS service centre). Sipcall is not a mobile operator, hence they don’t offer an SMSC service. I wrote a separate entry on the technical solution, the short version is – we have reverted to the old-fashioned solution of using a USB GSM modem with a pre-paid subscription.
With ISDN, you have the option of diverting calls. This is done in the ISDN exchange, not in our local Asterisk. When a call for a diverted number is received, the exchange is signalled to divert to another number. The advantage over rerouting with Asterisk is that the new recipient will see the original caller-id (instead of our own numbers).
I have not yet investigated how to do this with our new SIP/Sipcall setup.
Telefax is a very old-fashioned technology and we actually have very little use for it – nonetheless, our Asterisk setup is able to send and receive telefaxes, they are passed to Hylafax and routed to a recipient by email (with the fax attached in PDF format). We still receive faxes quite regularly, maybe once a week, although only spam and advertising. It’s been years since anybody sent a fax from our office.
I have not investigated this in detail, but according to various articles I have read, sending and receving faxes is not possible with SIP. I doubt it’ll be a great loss, and testing it with Sipcall is not a priority.