Migrating from PSTN to VoIP

So, for years… decades even, our telephone service has been via the Public Switched Telephone Network. Originally intended for just voice traffic, this later became our Internet connection, using dial-up modems, then using ADSL.

The number itself gained two digits in its life-time: originally 6 digits (late 70s/early 80s), it gained a 0 at some point, then in 1996 a 3 was prepended to all Brisbane numbers.

So yeah, we’ve had our phone a long time, and the underlying technology has remained largely the same for that time period. Even the handset is the same one from all those years ago. Telecom Australia used to pay for it as a “priority service” back then as my father was working for them at the time.

By September, this will change. The National Broadband Network is in our suburb, and a little while back I migrated the ADSL2+ connection over to HFC. After a brief hiccup getting OpenBSD talking to it, we were away. It’s been pretty stable so far. Stable enough now that I haven’t had the ADSL modem connected in weeks.

At the time of migration, I could have migrated the telephone too to NBN phone, however there’s a snag: how do you access NBN phone? The answer is the ISP sends you a pre-configured ATA+Router+WiFi AP (and sometimes there’s an ADSL modem in there too). As I had decided to use my own router, I didn’t have such a device.

I asked Internode about the SIP details, and the situation is this: when they provision a NBN connection, there’s an automated process that configures their VoIP service then stores the credentials and other settings in a ACS server. They have no visibility of these credentials at all. Ordinarily, if I had purchased hardware, they would have provisioned that with credentials that would allow it to authenticate over the TR069 protocol. So without spending $200 on a device I wasn’t going to use, I wasn’t getting these credentials.

There’s another option though: VoIP. The NBN phone is actually a VoIP service, so fundamentally nothing much changes. Keeping the services separate though does give me flexibility in the future. There’s a list of VoIP providers as long as my arm that I can go with.

VoIP is notoriously difficult to set up though, I didn’t want to mess up our only incoming telephone service, so I decided to migrate the NBN phone number over to Internode’s NodePhone VoIP service which would allow me to experiment.

Requirements

So, first thing to consider is what my needs are. We have 4 devices that are plugged into the telephone line at present:

• Telecom Australia Touchfone 200 wired telephone
• Telstra 9200a cordless telephone base-station
• Epson WF-7510 printer/scanner/fax
• Maestro Jetstream 56kbps modem

Now, we don’t send many faxes (maybe two a year), and the last time that modem was used, it was to dial into a weighbridge system in Rockhampton after my workplace had moved to VoIP.

That weighbridge system was originally built in 1995 on top of computers running SCO OpenServer 5 and using SCO UUCP over dial-up lines running at 1200 baud (some of the nodes were at remote sites with dodgy phone lines). In 2013 the core servers were upgraded with new hardware and Ubuntu 12.04LTS, but the UUCP links remained as the SCO boxes at sites were slowly replaced. mgetty would answer the phone, and certain user accounts would use uucico as the “shell”. For admin purposes, we could log in, and that would give us a BASH prompt.

Any time we had a support issue at work, muggins would be the one to literally “dial in” to that site. While it’s been years since I’ve needed to touch it (and I think now there’s a VPN to that site), I wanted to retain dial-up capability if needed.

As for the faxes… the stuff we’re doing can be done over email. Getting the modem and the fax machine working is a stretch goal.

99% of the traffic will be voice traffic. I’m not sure if it’s possible to use double-adaptors with an ATA, and so for hardware I opted for the Grandstream HT814. These are available domestically (e.g. MyITHub) for under AU$130 and looked to be pretty good bang-per-buck. I just wasn’t sure how well it’d get along with the old T200.

As a contingency plan, I also ordered an IP phone, a Grandstream GXP1615 which is also available locally. That way if the T200 gave me problems, I’d just wire up Ethernet to the old point where the T200 was and put the GXP1615 there.

Since I’ve got two SIP devices, I need to be able to control incoming and outgoing call flows. So that means running a soft-PBX somewhere. Asterisk is the obvious choice here, being a free-software VoIP package with lots of flexibility. OpenBSD 6.6 ships with Asterisk 16.6.2.

Opening the account

Opening the account is simple enough. As I was porting the NBN phone number over, I just needed some details off my last Internode bill. Later when I go to do the house phone some time next financial year, I’ll be after a Telstra bill (assuming Telstra Wholesale have sorted out their CoVID-19 issues by then).

I looked at the hardware options that Internode offered… and again, it was basically the same as what they offer for their Internet service.

One thing I note is that while Internode has been a great ISP (I switched to them in 2012), their credentials as a VSP are still developing somewhat.

Initial connection

Provisioning was much less smooth than my initial ADSL connection (which was practically seamless) or the subsequent move to HFC NBN. The first bump in the road was when they emailed me:

Subject: Internode: Your Internode NodePhone VoIP service xxxxxxxxxx can’t make/receive calls yet [xxxxxxxxx]

Following activation of your NBN service, we ran some tests to make sure things were running smoothly.

We weren’t able to detect that your NodePhone VoIP service (phone number xxxxxxxxxx) has been set up successfully.

Initial contact regarding the telephone service…

Okay, fair enough, on the date I received that email I had only just received the ATA that I had purchased (through another supplier). Evidently they thought I had the hardware already. I found some time and quickly cobbled together a set-up:

I did some quick research, rather than doing NAT, I figured siproxd was closer to my eventual goals, so I installed that on the border router as there were fewer knobs and dials to deal with. I configured the HT814 with the settings as best I understood them from Internode’s guides with one difference: I set the Proxy field to my border router’s internal IP address.

This failed with Internode’s server giving me a 404: Not Found response when my HT814 sent its REGISTER request. I took some captures using tshark and reported this back to their helpdesk. I disconnected the ATA since there was no sense in banging on their front door every 20 seconds: it wasn’t working.

They got back to me a few days later and told me I had the right user name, and that my number still wasn’t registered. Plugging the ATA in, same response, 404: Not Found. In exasperation, I tried some changes:

• Different settings on the HT814 (too many to list)
• Trying with Twinkle on my laptop connected to the DMZ, both through via siproxd and also shutting down siproxd and setting up NAT.
• Installing Asterisk on the border router (which was in the plans) and trying to set that up.

All three hit the same problem, 404: Not Found. I figured either I had entered the same wrong details 3 times, or it was definitely their end. So I reported that, unplugged the ATA and waited. This was on the 27th March.

On the 31st March, I get an email from Internode Provisioning to say they would be porting the number soon. After receiving this email, REGISTER worked, I was getting 200: OK in reply. Funnily enough, there was no challenge to credentials, it just saw my end and said “OK, you’re in”.

Calling outbound after this worked: the call trace showed the Internode end challenging the ATA for credentials, but once supplied, it connected the call, all worked. Inbound calls were a different matter though, a recorded (female) voice announced that the number was invalid. We were half-way there.

The next day, that message changed, it now was a recorded (male) voice announcing the number was unavailable, and offering to leave a voice-mail message. Progress. I had not changed my end, I still had T200 → HT814 → siproxd on the border router as my set-up. I was not seeing any incoming traffic being blocked by pf, although I was seeing people playing with sipvicious. I decided to firewall off my SIP ports, only exposing them to Internode’s SIP server.

Later on the help-desk got back to me. Despite their server telling me 200: OK when I sent a REGISTER, the number was still “not registered”. Confusing!

On a whim, I ripped out siproxd again and set up NAT on the border router. BINGO! We registered, and incoming calls worked. Internode use Broadsoft’s BroadWorks platform for their NodePhone VoIP service, and something about the operation of siproxd confused it. It then sent a misleading response leading my devices to think they were registered when they were not!

At this point, it was time to do two things: uninstall siproxd, and start reading up on Asterisk.

Configuring Asterisk

Asterisk is regarded as the “swiss army knife” of VoIP. It can be configured to do a lot. Officially, it is supported on Linux i386 and AMD64 platforms, but there are builds of it for platforms such as the Raspberry Pi.

OpenBSD also build and ship it in their ports. I wanted to avoid the pain of NAT, and I also wanted to avoid the telephone service going off-line if my cluster went haywire. So installation of this on the border router was a no-brainer. Yes, it’s adding a bit more attack surface to that box, but with appropriate firewalling rules, this could be managed.

As for configuration, there were two SIP channel drivers I could use, either the old chan_sip method, or the newer res_pjsip method. I had seen guides that discussed the older method (e.g. OCAU, WP), however in the back of my mind is the question: “how long do Digium plan to keep supporting two methods?” Thus from the outset I decided to use res_pjsip.

Audio CODECs

This is a pretty big aspect of configuration and should not be skimped on. You’ll find even if you buy all your equipment from one supplier, there are differences in what audio CODECs are supported. For instance the HT814 supports the OPUS CODEC (something I’d like to take advantage of eventually) but the GXP1615 does not. Some CODECs require patent licenses (e.g. SILK, SIREN7, G.722.1, G.722.2/AMR-WB), some did require patent licenses but are now “free” (G.729, G.723.1) and some are open-source (OPUS, Speex).

Some also go by multiple names. G.711a is also called PCMA and G.711u is PCMU. Pretty much everything supports these, and depending on the country you’re in, one will be preferred over the other. In my case, G.711a is the preferred option.

Your voice provider is a factor here too. Some only support particular CODECs, some will allow any CODEC. NodePhone VoIP allegedly will allow you to use whatever you like, but calls into and out of their network to non-VoIP targets may be restricted to narrow-band CODECs.

Internode-specific gotcha: G.729

One gotcha I stumbled on the hard way was that sometimes SIP implementations do not play by the rules.

Specifically, I found once I got Asterisk installed, incoming calls would be immediately hung-up on the moment I answered. Mobile or PSTN didn’t matter. I fired up tshark again and dug into the problem. The only clue I had was this error message:

[Apr  5 16:27:40] WARNING[-1][C-00000001] channel.c: Unable to find a codec translation path: (g729) -> (alaw)

Even if I specified only use G.711a (alaw), somehow I’d still get that message. I asked about it on the Asterisk forum. I was seeing this pattern in my SIP traffic:

|Time     | ${PROVIDER}                           |
|         |                   | ${ASTERISK_BOX}   |                   
|0.000000 |         INVITE SDP (g711A g7          |SIP INVITE From: <sip:${MYPSTNNUM}@${PROVIDER}29;user=phone> To:"${PROVIDER_NAME}"<sip:${MYSIPNUM}@${PROVIDER_DOMAIN}> Call-ID:BW061858648050420-1965318496@${PROVIDER}   CSeq:612303213
|         |(5060)   ------------------>  (5060)   |
|0.003443 |         100 Trying|                   |SIP Status 100 Trying
|         |(5060)   <------------------  (5060)   |
|0.025416 |         180 Ringing                   |SIP Status 180 Ringing
|         |(5060)   <------------------  (5060)   |
|0.954015 |         200 OK SDP (g711A g7          |SIP Status 200 OK
|         |(5060)   <------------------  (5060)   |
|0.986287 |         RTP (g711A)                   |RTP, 6 packets. Duration: 0.099s SSRC: 0x4F53507
|         |(27642)  <------------------  (18024)  |
|1.092729 |         RTP (g729)                    |RTP, 3 packets. Duration: 0.041s SSRC: 0xCD74F85
|         |(27642)  ------------------>  (18024)  |
|1.097523 |         ACK       |                   |SIP Request INVITE ACK 200 CSeq:612303213
|         |(5060)   ------------------>  (5060)   |
|1.099552 |         BYE       |                   |SIP Request BYE CSeq:29850
|         |(5060)   <------------------  (5060)   |
|1.149521 |         200 OK    |                   |SIP Status 200 OK
|         |(5060)   ------------------>  (5060)   |

I was reliably informed that this was definitely against the rules. So another help-desk email informing them of the problem. If you’re an Internode NodePhone VoIP customer, and you see the above behaviour, these are your options:

1. Purchase and Install Digium’s G.729 CODEC: only an option if you are running Asterisk on a i386 or AMD64-based Linux machine.
2. If, like me, you’re running Asterisk on something else, or you despise proprietary software, there is an open-source G.729 CODEC. On OpenBSD, install the asterisk-g729 package.
3. Asterisk have added a work-around in later versions of their code. Patches exist for version 17, 16 and 13. It is also included in 13.32.0, 16.9.0 and 17.3.0. OpenBSD 6.7 will likely ship with a version of Asterisk that includes this work-around.
4. You can also complain to their help-desk. We can work-around their problem, but really, it’s their end that’s doing the wrong thing, they should fix it.

Dial Plans

The other big thing to consider is your dial-plan layout. Every SIP endpoint is assigned a context which is used in extensions.conf to determine what is meant when a particular sequence of digits is entered or how a call should be routed.

The pattern syntax is documented on their Pattern Matching page. Notably, extensions are “literal” if they do not begin with an underscore (_) and a X matches any numeric digit. So an example:

exten => 123456,1,DoSomething()
exten => 123987,1,DoSomethingDifferent()
exten => _123XXX,1,DoSomethingElse()

The non-pattern extensions take precedence. So the number 123456 would exactly match the first extension listed above and would call DoSomething(). The number 123987 would exactly match the second entry and would call DoSomethingDifferent(). Any other 6-digit number beginning with 123 would trigger DoSomethingElse().

Avoiding expensive mistakes

Before doing anything, it’s worth reading Asterisk’s page on Dial plan Security. You do NOT want someone to call your PBX, then dial outbound to expensive international numbers and run up a big phone bill! In particular, you want to keep the default context as lean as possible! It’s fine to put all your internal extensions in default, but anything that dials outside, should be in a separate context for internal extensions.

Dialling more than one phone at a time

It is possible to have a dial-plan entry ring multiple phones in parallel by separating the endpoints with & symbols, but don’t put spaces around the & characters! So Dial(PJSIP/ep1&PJSIP/ep2&PJSIP/ep3), not Dial(PJSIP/ep1 & PJSIP…). The most obvious use case for this is when someone rings the home number and you want all phones to ring.

Incoming calls in the dial plan

When a call comes in from outside, the number dialled by the outside party (i.e. your number) appears as the extension dialled.

A simple option is to just ring everyone… so if your phone number was, say 0735359696, you’d create a context in your extensions.conf like this:

; Incoming calls
[incoming]
exten => 0735359696,1,Dial(PJSIP/ep1&PJSIP/ep2&…)

… then in pjsip.conf, assign that context to the endpoint:

[Provider-endpoint]
type=endpoint
context = incoming
; … etc

Outgoing calls

Usually you want to be able to ring outbound too. Some guides suggested the pattern _X! can be used to “match all” but I couldn’t get this to work.

Fax over IP

I mentioned that we had a fax machine we wanted to continue using. Whilst we don’t use it every day, it’s nice to know it is there if we want to use it.

Likewise with the dial-up modem. Even if I needed to do reduced-speed for it to work, it’d be better than nothing for situations where I need to use dial-up. It’ll never connect to the Internet again, but that doesn’t make it useless.

There are two ways to do Fax over IP. One is to use G.711u and hope for the best. The other is to use T.38, where the ATA basically “spoofs” the fax modem at the remote end, decodes the symbols being sent back to digital data, encodes that in UDP packets and transmits those over the Internet. The other end then reverses the process.

The good news is there are diagnostic tools out there for testing purposes. You don’t (yet) have to know of someone who has a working fax. Here in Australia, there’s FOLDS-B.

I’d recommend if you’ve got multiple fax-capable devices though, you plug two of them into separate ports on one or more ATAs and try faxing from one extension to the other. I tried calling FOLDS-B directly at first with no luck, then tried setting up Asterisk with an extension that calls SendFax to send a TIFF image, but had no luck — handshaking would be cut short after a second.

Using two internal endpoints saved a lot of phone calls externally and allowed me to “prove” the ATA could work for this.

Once you’ve got things working locally, you’re ready to hit the outside world. You’ll need a fairly complex image to send as it needs to be transmitting for ~70 seconds. I tried a few pages (e.g. this one) on the flatbed scanner of the WF-7510 without much luck… the fax would barely muster 20 seconds of data for them even at “fine” levels.

I had better luck using the 56kbps modem and a copy of efax-gtk. I took RCA’s test card image off Wikipedia as a SVG, loaded that up into The Gimp, rendering the SVG at 600dpi, extending the image size to A4-paper sized, rotated it to portrait mode, then converted it to 1-bit-per-pixel monochrome with patterned dithering and saved it as an uncompressed TIFF.

I then passed this through tiff2ps which gave me this file:

Sending that at 14400bps took 72 seconds. Perfect! Then the reply came back. Fax reception is very much a work-in-progress, so rather than receive on the VoIP line, I decided to use the PSTN to receive the reports. This was accomplished by specifying my PSTN phone number in the fax identity (FOLDS-B evidently doesn’t check this against caller ID).

I tried again, and this time, received the report. FOLDS-B got a garbled mess apparently. The suggestion was to set the speed to 4800bps. In efax this is accomplished by setting the modem capabilities:

CAPABILITIES
       The capabilities of the local hardware and software can be set using a string of 8 digits separated by commas:

       vr,br,wd,ln,df,ec,bf,st

       where:

       vr  (vertical resolution) =
                0 for 98 lines per inch
                1 for 196 lpi

       br  (bit rate) =
                0 for 2400 bps
                1 for 4800
                2 for 7200
                3 for 9600
                4 for 12000 (V.17)
                5 for 14400 (V.17)

So in this case, I should set the capabilities to 1,1,0,2,0,0,0,0. I tried this, but then found the call just didn’t negotiate either. On a whim again I tried 7200bps (that’s 1,2,0,2,0,0,0,0). Eureka! I got this:

The transmission level and SNR are pretty much spot-on. Emboldened by this, I tried again faxing the same image (which I had printed out) from the WF-7510. I chose “Photo” quality and scanned it on the flat-bed. This didn’t work so well — evidently some detail was missed and it only transmitted for 60 seconds so I tried again, and faxed something else for the second sheet.

FOLDS-B wasn’t happy about the second page, but at least it had something to go on. The fax modem in the WF-7510 doesn’t appear to have a speed control other than turning V.34 mode (33.6kbps) on/off. It appears it tried sending at 14400bps. FOLDS-B tells a sorry tale:

Now it is possible to get near perfect results at 14400bps through an ATA that supports T.38. Maybe a longer transmission on the first page might have helped, but fair to say the speed is not helping matters.

Transmit level is very quiet, and I can’t see a way to adjust this on the WF-7510. Nor can I force it to V.29 (7200bps) mode.

There’s allegedly a “reference” test sheet you can get by “receive polling” 019725112. When I ring this number (with a fax or telephone) I get a recorded message saying the number is not valid. If anyone knows what the correct number is, or how to get a copy of this reference sheet, it’d be greatly appreciated.

So yes, Fax over IP is doable… a lot of pissing about with ATA settings… and you better hope the fax machine itself has lots of knobs and dials. Hylafax + t38modem will probably crack this nut. Then again, so will email.

Configuration Settings

Asterisk Configuration

So whilst my set-up is still a work-in-progress, I figured I’d post what I have here.

Most of these are defaults shipped with OpenBSD 6.6. Specifically of interest would be pjsip.conf and extensions.conf.

pf firewall rules

You’re going to want to pierce your firewall appropriately to expose yourself just enough for your VoIP service to work.

# Define some interfaces
internal=em0

# SIP addresses
sip_provider=203.2.134.1  # sip.internode.on.net
sip_endpoints="{ 10.0.0.3, 10.0.0.4 }"

# Allow SIP from router to SIP devices/softphones and vice versa
pass in on $internal proto udp from $sip_endpoints to self port { 5060, 10000:30000 }
pass out on $internal proto udp from self port { 5060, 10000:30000 } to $sip_endpoints

# Allow SIP from Internode to self and vice versa.  This could be
# tightened up a lot further.  Maybe try some calls both ways and log
# the traffic to see which specific ports.  "All UDP ports" is in line
# with Internode recommendations:
# https://www.internode.on.net/support/faq/phone_and_voip/nodephone/troubleshooting_nodephone/#Do_I_have_to_open_up_any_ports_i
pass in on egress inet proto udp from $sip_provider to self
pass out on egress inet proto udp from self to $sip_provider

Grandstream HT814 settings

Some notable settings first before I dump full screenshots…

Ring Cadence Settings

A special thank-you to @scottsip on the Grandstream forums for pointing me to this document from the ITU (Page 4 has the settings for Australia). Also worth mentioning is this Whirlpool forum thread and this review/teardown of the Grandstream HT802. The ones marked (?) I have no idea what they’re for.

• System Ring Cadence: c=400/200-400/2000;
• Dial Tone: f1=400@-12,f2=425@-12;
• Busy Tone: f1=425,c=38/38;
• Reorder Tone: f1=480,f2=620,c=25/25; (?)
• Confirmation Tone: f1=350@-11,f2=440@-11,c=100/100-100/100-100/100; (?)
• Call Waiting Tone: f1=425,c=20/20-20/440;
• Prompt Tone: f1=350@-17,f2=440@-17,c=0/0; (?)
• Conference Party Hangup Tone: f1=425@-15,c=600/600;

Notable fax-related settings

I changed lots of these trying to get something working, so if you set this and still don’t get any joy, check the screenshots below.

• Fax Mode: T.38
• Re-INVITE After Fax Tone Detected: Enabled
• Jitter Buffer Type: Fixed
• Jitter Buffer Length: Low (note, I can get away with this because it’s Ethernet LAN from ATA to Asterisk box)
• Gain:
  • TX: 0dB
  • RX: 0dB
• Disable Line Echo Canceller: Yes
• Disable Network Echo Suppressor: Yes

Screenshots

• 

• 

Grandstream GXP1615 Settings

These are much the same as the HT814 above. The cadence settings use a slightly different syntax (they don’t have the @nn parts) and there are more tone settings. Again, the ones marked with (?) have an unknown purpose.

• System Ringtone: c=400/200-400/2000;
• Dial Tone: f1=400,f2=425;
• Second Dial Tone: f1=450,f2=425; (?)
• Message Waiting: f1=525;
• Ring Back Tone: f1=1209,f2=852,c=20/20-20/200;
• Call-Waiting Tone: f1=425,c=20/20-20/440;
  • Gain: Low
• Busy Tone: f1=425,c=38/38;
• Reorder Tone: f1=480,f2=620,c=25/25; (?)

The steps forward

Things I need to figure out with this system:

• Feature Codes: these are used to signal events like “transferring calls”, “park”, and other features you might want to initiate in the PBX. They are normally signalled using DTMF tone sequences, however this can give problems if your tones clash with some IVR you’re interacting with. I’m currently researching this area.
• OPUS support: I mentioned the HT814 supports it, and as it’s an open CODEC I’d like to support it. There is this project which provides OPUS support to Asterisk. I’ll probably look at writing an OpenBSD port for it.
• Voice mail and IVR… I’m thinking something along the lines of “Dial the year of birth of the person you wish to reach”… then that can ring the relevant phones or offer to leave a message.
• I’d like to test wideband audio at some point. Work seems to have G.722 on their phones (or maybe its OPUS?), I should see if wideband pass-through in fact works.

Tags: asteriskfaxfax-over-ipgrandstream-ht814internode-nodephoneopenbsdopenbsd-pft.38voip