VOIP Tech Chat → PBXes.com (Free): Can't get Inbound Route working correctly

I've been playing with PBXes.com over the past several days and have everything working to my satisfaction with one exception: Inbound Routing. Anyone in here using PBXes who can guide me? The free service is unsupported -- you can't even post in the PBXes forum; the FAQs are far too general; and the PBXes Wiki has never been finished.

I have, without trouble, set up a number of my SIP providers (Callcentric; OneSuite; Gizmo5) as trunks in PBXes. (One exception to "trouble-free": IPComms. I can register to it just fine with a Zoiper softphone, but trunk registration typically fails or doesn't "stick" in PBXes. That's another post I guess...)

I have, without trouble, set up both lines of my PAP2T as extensions (called "100" and "200") to PBXes, and have also set up my Zoiper softphone as extension "300" (no quotes). All works as expected.

I have, without trouble, set up Outbound Routing in PBXes. Using very simple rules (I'll get more fancy later on) all calls prefixed with "1777" go out over Callcentric (so as to complete free SIP calls to CC subscribers); all calls prefixed with "1747" go out over Gizmo5 (so as to complete free SIP calls to G5 subscribers); and all other calls go out over OneSuite. All works as expected based on the numbers dialled.

However, I can't seem to get Inbound Routing working the way I want it to work, or the way I think it should work. The PBXes interface for Inbound Routing is very simple. You can identify a trunk or a CID, and then you can set as a destination a particular extension based on those variables. If I omit both trunk and CID, I get a "general" inbound route that does indeed route all calls to the one identified extension. If I add a CID but no trunk I get a CID-specific route that does indeed route to a specific extension based on the CID. However, I can't seem to route to a specific extension based on incoming trunk. And that's what I want to accomplish.

The mouse-over text help for "trunk" in the Inbound Routing dialog says: "Define the expected trunk on incoming calls. For most providers this is the trunk name, but for some it's the username of the trunk. Leave this blank to match calls from any trunk."

No matter what I have tried, I can't get trunk-specific routing to an identified extension. For example, let's say I want all inbound OneSuite DID calls to go to extension 200. I created an inbound route with the "trunk" identified as "OneSuite" (no quotes; what I named the trunk when I set it up, and a name that works just fine for Outbound Routing) and selected extension 200. If I have no "general" inbound fallover route, an inbound call to the OneSuite DID fails. The caller hears a short ring, silence, and a hangup. The call is never delivered to an extension, although logs establish that the call is indeed getting into PBXes. If I do have a "general" inbound route, the call is delivered, but to the "general" route's selected exstension (100, rather than 200). I have tried any manner of naming the "trunk" in the Inbound Routing dialog, but nothing has worked. I have named the trunk by its DID in the following formats: 1773XXXYYYY and 773XXXYYYY. I tried to name it "+1773XXXYYYY" (no quotes) but PBXes complained about the "+". I tried to name it "OneSuite" (including the quotes) but PBXes complained that "a route for this DID/CID already exists." It doesn't exist, of course. I'd thought that the syntax "ABC" might be interpreted as null, and therefore PBXes was complaining that a no-trunk name, no CID name inbound route already existed in the form of the "general" route, but this error mesage is displayed even if there isn't a "general" route.

Remembering that the mouse-over text help says: "For most providers the expected name for the trunk is the trunk name, but for some it's the username of the trunk", I'm thinking I have to use the "username" of the trunk, but I have no idea what that is for this particular OneSuite trunk. I tried my user name for my OneSuite account, but that didn't work.

My Asterisk-savy friend thinks the problem is simply that the free version of PBXes doesn't permit more than one Inbound Route, even though I can't find that limitation spelled out anywhere, and even though the inbound route dialog clearly invites one to select more than one.

Thoughts, oh VoIP experts?

You can have as many inbound routes as you want.

Check if you have any kind of sub-accounts at your provider, possibly assigned per DID.

Also, what do you see in the call history when call comes to the generic route? It could give a clue.

I use none of your specified providers, so I can't vouch for them. But for mine, I use just the trunk name (careful of caps) to direct all of that providers inbound traffic to a chosen destination without problems. I generally force regular hours by default.

Well, it's not the providers because if I eliminate PBXes.com from the picture and register to any of them directly using my PAP2T, I have no problems at all. The trouble clearly lies with the inbound routing that I am setting up. All the SIP trunks work fine with PBXes -- terminating at extension 100 -- if I set up that general inbound route. I just can't figure out how to get inbound traffic on specific trunks to go to extension 200.

Following your suggestion, I eliminated caps from all of my trunk names. Changed nothing. I also force regular hours. Changed nothing. Having a "*" in the "hours" and "days" field also changes nothing compared to leaving those fields blank (which is what I had been doing.)

My guess is that the trunk name is somehow not correct (but, what then should it be?) or I have some syntax error that I can't see.

How does what I posted differ from your settings?

It is provider... To give you an example, while most of providers work with just a trunk name, for CWU trunk name was their subaccount; for voip.ms it was the DID, etc.

The key part here is to determine how provider identifies DID internally - as if you had multiple DIDs on a single registered trunk. I couldn't figure it out with voip.ms until I added a second DID and then saw the differences how they presented within their system ("account/DID" meant DID is the identifier.)

When I said "it's not the provider," I mean it's not their service or reliability. The providers are all up and working fine. The problem is my not knowing the correct name to use for the inbound trunk.

So, if the "key part here is to determine how the provider identifies the DID internally," how do you find out that information without having to purchase a second DID as was your solution?

have you tried using the username for the provider in the 'trunk' field for the inbound route?

Yeah; that was the very first thing I tried. And, after vastly more screwing around, it turns out that therein lies the problem with this provider. (There are other problems with other providers, but those are other stories).

This one particular provider, OneSuite, used to use a simple, short, username/auth name for softphone/ATA SIP registration. In my case, it was the username that I initially selected on sign-up with the service several years ago: a 7-letter word followed by 4 numbers.

This past May, for softphone/ATA registration with their service, OneSuite required all users to append

"-voip.onesuite.com"

after the existing username/auth name. So, my new username/auth name became:

"[original7-letter-word-and 4 numbers]-voip.onesuite.com".

I think new, long user/auth name is what correctly should be inserted into the "trunk" field when creating an inbound route in PBXes. But, PBXes barfs when I try to do so, failing with the message "name too long" or something like that. (I assume PBXes.com is a web GUI front-end to an underlying Asterisk set-up). So, I can't select an inbound route in PBXes.com for OneSuite because the correct trunk name is disallowed as too long.

This long username/authname has proved to be a problem before. I think I initially tried to use OneSuite as the SIP-forward-to provider for IPKall DID completion. IPKall similarly complained that the username was too long.

I've tested this by using borrowed credentials for a VoIP provider (Voicepulse or VoIPStreet; note sure which right now) where the DID was also correctly the "trunk" for PBXes.com inbound route purposes. Everything worked as was expected: the IB call was routed to the non-default extension.

Moral of the story, as has been suggested above: it can be hard to determine the exact name of the actual trunk for purposes of directing IB traffic to extensions via IB routes. And, sometimes even when you have the correct name for the trunk, it is non-standard in length and PBXes.com (and possibly even Asterisk) will barf and reject same.

Thanks all.

I have been using PBXes.com extensively for last a year and half. Although I have an account with PBXes.com for over 3 years but at the beginning, there was bandwidth issue so I had to stop using PBXes.com for almost two years. Currently, I am using PBXes.com to have 6 DIDs from IPKall and 1 DID from IPcomms forward to my SIP URI at PBXes.com. Just as you have noted, PBXes.com cannot register with IPcomms. Furthermore, I have 1 DID from messagenet, 1 DID from OneSuite, 1 DID from sipgate and 1 DID from VoxOx register via PBXes.com as trunks. Additional trucks register via PBXes.com are SIPphone (Gizmo5), SIPsorcery (just for Google Voice dial out), swiftvox (VoIP.ms, just as secondary dial out, after Google Voice), VoicePulse (as back-up dial out, after swiftvox or VoIP.ms).

Since I want most of DIDs including one from OneSuite to ring all phones, I did not aware the issue you detailed in this thread. After reading entire thread, I tried your configuration and got same result as you.

On the other hand, have you tried to set general (catchall) inbound route to extension 200 then configure trunks other than OneSuite to extension other than 200 instead?

I never use PBXes.com but have you tried changing the Onesuite voip.onesuite.com to IP address 64.77.227.110 I think you'll save about 4 characters not sure if that will help though.

Hi, Hok; I've seen your posts elsewhere in this group.So, are you saying that you can get your IPComms DID to terminate, inbound, into PBXes.com? That you can call your IPComms DID, get the call into PBXes.com, and from there to an extension registering to PBXes? That might be a way around the inability to get IPcomms to register as a trunk. Would you explain how you configure the IPcomms SIP-forwarding to the SIP URI at PBXes.com

(Interestingly, I've compared the Asterisk source from PBXes to that suggested by IPComms and the code is for the most part the same. There are differences, but I don't see one that is significant or that would prevent registration. In fact, the registration line that is in my PBXes source is identical to that outlined by IPComms.)That's a potential work-around that I have not tried; thanks. Good suggestion. That would work if indeed all of the other trunks can in fact be selectively inbound routed to 100. But I think some of the other trunks act glitchy, also. I'll check.

No, I haven't tried that. Not sure if it would work because that's a general end-point IP and I'd guess I need a trunk name specific to my registration. But it's certainly easy to check out; thanks for the suggestion.

@brg

Have you tried it? I've tried it on X-lite and it didn't work. I think part of the Onesuite VoIP username can not be change to IP address (the voip.onesuite.com)

I get 404 user uknown error.

Tell us how it went and if you found any solution.

Hi Zaldy:

It's two issues:

The "username-voip.onesuite.com" is essential to registration and probably couldn't be changed so as to replace the words with the IP. So yea, that's why it wouldn't work on registration with X-lite.

For purposes of what I was trying to do, creating an inbound routing for my OneSuite SIP trunk via PBXes.com, naming the trunk "username-64.77.227.110" is still "too long." Oh well...

YesYesPlease see threads with following URLs:
»Re: IPComms Free DIDs now with sip registration maybe??
»Re: IPComms Free DIDs now with sip registration maybe??

I have IPComms configured for SIP. I have it working fine registering to my PAP2T and to various softphones.

The problem that Hok and I were discussing is the fact that even though the both of us have IPComms configured for SIP termination, and even though it works fine from an ATA, we can't get it to register with PBXes.com. I was asking Hok to explain his apparent work-around to the problem.

I'll reread the links you directed me to (which links do not mention PBXes.com) to see if there is something that answers my questions about IPComms registering with PBXes.com. Maybe what you are saying is that the work-around I asked about is discussed in those threads.

Edit: OK; I was being dense. I now understand the fix -- discussed in the context of SipSorcery -- and will see if I can use it to get things workinw with PBXes. Thanks.

Sorry for being not much of a help

I got an idea though. You said your username is 11 characters long already. Well if you can short it out to 3 characters then maybe it will be short enough to work.

I think you need to sign up again though for a new Onesuite account and just transfer your balance and your VoIP number from your current one to the new one. Just call Onesuite support if this is feasible.

No worries! Maybe I'll check to see if I can do what you suggested, but more likely that will end up being too much of a bother. Thanks for the idea, though!

Hi all,

Would I get better or worse or the same voice quality (in/out) when I click Yes Bypass Audio ?

OK, I know it's obvious it's better or equal but I just want to be sure

Anyone really did a comparative study ?

When setting "audio bypass:" to yes be sure to make some test calls. Audio bypass transmits audio directly between phone and provider if both are enabled without going through PBXes. Few devices and providers do not support it resulting in dropped calls or missing audio. Of course, call recording will not work anymore as audio did not go through PBXes.

In other words, you have to try your own as everyone has different device and/or provider.

By the way, for following trunks, I have set audio bypass:
• Google Voice via SIP sorcery
• Gizmo5 for calling toll-free numbers and Google Voice incoming calls SIP connection
• demo.switchvox.com
• Ribbit Mobile via SIP sorcery

I do not notice much voice quality improvement, but may just a bit better. However, I am sure it does save PBXes bandwidth.

Added my tests.
audio bypass in my pbxes setup. I am using spa1001.
IPKall
SipGate

No audio degrading was observed.

btw, hok, are you using a paid pbxes account? you have so many trunks (obviously more the five).
I am thinking to add my sipsorcery into pbxes trunks. sipsorcery is not very stable, comparing to pbxes and voxalot. My plan is to use ss for outbound only. while let pbxes and voxalot to handle inbound calls. My inbound was using ss and seemed too many troubles.

Actually, virtually all trunks can be set to audio bypass due to no VSP is behind NAT. On the other hand, almost no extension can be configured to audio bypass due to almost all extensions are behind NAT. If audio bypass is set for extension, it will ring but no audio.

While sipgate requires registration that has a trunk, IPkall is forwarded to PBXes SIP URI that should have no trunk. How did you set audio bypass for IPKall?

With PBXes, SIP sorcery is required for outbound and inbound calling of Ribbit Mobile, and is required for outbound calling only of Google Voice. However, you can save a trunk if you also let SIP sorcery to handle Google Voice inbound calls. All other inbound calls are handled by PBXes only.

SIP sorcery has backup server now. They are a lot more reliable than before. However, you are correct; SIP sorcery is still not as reliable as PBXes even just using one of PBXes servers. Why you still have the need to use Voxalot after using PBXes?

If I remember correctly, PBXes started limit free accounts to have five trunks about a year ago. People with free account before the limit imposed had more than five trunks can continue to keep their extra number of trunks until they delete those trunks. Once those trunks are deleted, they cannot be added back if they already have five trunks. In contrast, people can maintain same number of trunks and just updating trunks configuration when needed.

ipkall is one of my pbxes trunks (at least, I did set up and it works). it is not registered within pbxes (no need). I use it for inbound call (three IPKall numbers) and toll-free outbound calls.
sipgate is only one registered in my pbxes. it provides in/out-bound call. it won't work with voxalot, a reason I use pbxes (voxalot not very useful).
I tried sipsorcery registered in pbxes. It worked fine but I didn't decide if I should move to this direction.
Reliability is very important in phone service. it's my experience since I have used sipsorcery. It's safer to keep two lines on my ata/spa1001. ss-->line1. pbxes-->line2.
My pbxes account is basic/free, limited up to five trunks. To save bandwidth, yes I bypass all audio (sipgate, ipkall as well as voxalot). voxalot has a unique feature: no betamax FUP. Otherwise, it could be dropped as pbxes is more useful and powerful.

So, based on what I have read above, it appears that Sipgate and Gizmo5 trunks could safely be set in PBXes.com to "audio bypass" without problem, correct? I will try that. If folks have experienced trouble doing this, please let me know.

In my case, my IPKall number is SIP-forwarded to my free Callcentric account SIP number, and the Callcentric SIP number is in turn registered with PBXes. Based on the above, it looks as if it would be OK to also set the Callcentric trunk to "audio bypass," yes?

Again, it is safe to set all trunks to have audio bypass.

I just opened a ribbit account last weekend. not activated yet.
need to know how to add SIP phone.
The prompt says to send an email to ribbit.

Post by ribbit web
Please note: Effective April 29, 2010, the ability to provision a new X-lite SIP phone to your account was removed. X-lite SIP phones provisioned before this date remain in service.

If your SIP phone (installed before April 29, 2010) stopped working, it might be due to improper password settings. To maintain maximum security in the network, our network requires SIP devices to use the SAME password used when setting up the device in your Ribbit Mobile online account.

Please follow these password setup steps, and, your SIP phone should be restored:

brg,

yes. sipgate has no problem using audio bypass in my pbxes account.

I tried sipsorcery, F9, sipgate and IPKall. I am using sipagte and IPKall. so far so good. F9 has recent carrier change. so I don't it but it was working.

If I remember correctly, activation in this case means forward all calls received by your mobile phone to Ribbit Mobile, then you are correct; you do not have to do the activation. To obtain a uxxxxxxx SIP log in user name, just add a SIP phone.

I guess I am a little late on ribbit play. ribbit doesn't offer IP/SIP setting in the old way any more. It asks me to send an email to them. I installed X-lite and followed its previous manual, tried hours.

Anyway, I sent an email to them. Let's see.