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Unread 09-09-2008, 05:24 AM   #1
olegp
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Default forwarding Sipbroker call

I am trying to get maximum usability from Linksys PAP2T.

The phone connected to line 1 and it configured with Voipstunt for outgoing calls. Username here has to be __alphabetical__ (something like abcdef@voipstunt.com).

I want to receive incoming calls on the same phone. Sipbroker seems to be a perfect candidate, but it uses __numerical__ username for the destination (something like 123456@yourprovider.com) – can NOT be combined with Voipstunt on line 1.

All right, I have second line in PAP2T – configure it to receive incoming calls without registration, username 123456, setup Sipbroker destination 123456@myhost.com – it works, I can call my Sipbroker alias from SIP phone (X-Lite) or through Sipbroker PSTN phone number. But I am receiving call to Line 2 – have to have second phone… Not convenient, I want to use only ONE phone!

Now I configure PAP2T Line 2 to forward incoming calls (after a few seconds) to Line 1. Calling my Sipbroker alias from X-Lite – it works. First it rings phone on line 2, than rings line 1, I can answer and talk… Calling the same Sipbroker alias through PSTN number – line 2 rings for a few seconds (I can answer and talk), but line 1 does NOT ring - as soon as ATA attempts to forward call to line 1 caller hears busy beeps...

What’s wrong? The same setup does work from soft-phone (X-Lite) and does not work through Sipbroker PSTN. Does Sipbroker prohibit forwarding? Why?

Is there a way to setup Sipbroker destination with alphabetical username (user@provider.com)? Any other ideas? Almost sure that Asterisk may solve the problem, but I would prefer to avoid unnecessary complexity.

Thank you for reading my long description.
Any suggestions would be appreciated.

--- oleg

P.S.
Just recognized – Sipbroker allows alphabetic username in destination, but does NOT allow port (user@provider.com:5061). Although the listing on SIPBroker - Provider White Pages includes one record with port (proxy.netphonedirectory.org:5065). How to specify port?

Last edited by emoci; 09-12-2008 at 04:32 AM.
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Unread 09-12-2008, 04:33 AM   #2
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The ideal situation would be to send calls to Line 1 directly...

Now, when you say MyHost.com above, does this mean that you have another provider set up in line 2, or are you using a service like DynDNS or No-IP, or do you happen to have a static IP...

The issue is SipBroker does not play well with most BetaMax providers (VoipStunt included) but you can assign yourself an Alias on any other domain...

The easier solution:
1. Get a Free VoXalot acct.
2. Register your ATA with VoXalot
3. VoipStunt can be configured within VoXalot, so although your ATA registers to VoXalot, you'll still be able to make outgoing calls via VoipStunt (think of VoXalot like a hosted version of an ATA)
4. At this point anyone can call you via SipBroker by dialing *010123456 (of course instead of 123456 it'll be your own 6 digit VoXalot number)

Last edited by emoci; 09-12-2008 at 04:40 AM.
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Unread 10-10-2008, 12:24 AM   #3
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Hi emoci, thank you for the response. Sorry for my delayed reply – the forum did not show my original post for couple of days – I guessed it was censored (although I could not imagine a reason) and gave up - did not open forum until now.

Answering your question – I control DNS myself (on third party servers) – pretty much the same as DynDNS. I was trying to setup Line 2 to receive direct IP calls.

Solution with VoXalot should work, but it introduces dependency on third party service which would be nice to avoid.

Meantime another glitch – calling from my cell to Sipbroker PSTN – hear the answer, can call my alias, it connects. Now calling from Packet8 phone to Sipbroker PSTN – hear the answer, dialing the alias – hear busy signal. At that my ATA does not receive any SIP request. Tried to call other numbers via Sipbroker PSTN – same story – can call from cell, but can not call from Packet8. Any explanation?

---oleg
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Unread 10-10-2008, 12:55 AM   #4
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Quote:
Meantime another glitch – calling from my cell to Sipbroker PSTN – hear the answer, can call my alias, it connects. Now calling from Packet8 phone to Sipbroker PSTN – hear the answer, dialing the alias – hear busy signal. At that my ATA does not receive any SIP request. Tried to call other numbers via Sipbroker PSTN – same story – can call from cell, but can not call from Packet8. Any explanation?

---oleg
Ok,

1. New users posts are automatically moderated when they post links, hence why it took a while to see your post

2. One of the main reasons to bring VoXalot into play is to avoid forwarding from one line to the next

3. I'm gonna try explaining a bit of the background on SipBroker, but keep in mind I am not 100% clear on it either.

A good chunk of the PSTN access numbers have been set to use re-invites. As a method this ensures more users can be accomodated.

It however uses a method where SipBroker stands in the middle sending out messages to both ends of a call on how to reach each other, but does not get involved in the actual audio stream, so the two ends have a direct connection to each other for audio.

That said when the calling party is using a VoIP line it is possible that it has compatibility issues with Re-Invites and hence your trouble...

When you place a call via SipBroker the common path is:

PSTN Caller>> Provider of PSTN Number (eg. CallCentric) >>SipBroker Server >> Network You're Calling>> Called User's Device

Re-invites make it possible that despite that long path audio is actually just:
PSTN Caller (which is assigned a temporary adress by the SipBroker Server)>>Called User's Device

Now in your situation the path would look like:

Packet8>> Packet8 Termination Provider>> PSTN>>Provider of PSTN Number (eg. CallCentric) >>SipBroker Server >> Network You're Calling>> Called User's Device

Re-Invites will attempt to reduce that to:
Packet8>>Called User's Device
for audio purposes.

If Packet8 or the Packet8 termination provider do not support Re-Invites, you may run into trouble.

Unfortunately the SipBroker PSTN access numbers were meant to be used from the PSTN. It is true they do work fine when called with most VoIP lines as well but incompatibilites can lead to issues...

4. Alternatively it could also be simply a DTMF issue where the device/softphone you have with Packet8 is not sending digit tones properly

One thing I would try for now, is to add a # to the end of the dialed number (So *010123456#).

If is it the case I described in (3) there isn't much that we can do. Question though: Does Packet8 allow direct SIP URI calls? (that could a solution)
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Unread 10-10-2008, 01:43 AM   #5
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Quote:
Originally Posted by emoci View Post
Question though: Does Packet8 allow direct SIP URI calls? (that could a solution)
Last I heard, no. But supposedly FWD ("Free World Dialup") has a P8 gateway that can be used to call P8 numbers via SIP URI. So if you are FWD user, you could (in theory) forward to FWD and have FWD forward to P8.

Of course, FWD now has an annual fee for their service, and using FWD still requires you to get a 3rd party service (FWD) involved. So I'm not sure you gain much by this approach. Still the FWD to P8 gateway (if it's still up), is one potential way to call a P8 number via SIP URI...
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Unread 10-17-2008, 04:42 AM   #6
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Hi Emoci, DracoFelis, thank you for responses.
Regarding Packet8 – they do not seem to be open to any interfacing. There are some old success stories in forums, but nothing seems to be working now. Although I will check FWD again.

I tried to use # at the end of dialed number. It did not help. BTW, I've discovered that some SipBroker PSTN numbers work (with or without #), while other do not work (regardless of #) when calling from Packet8 phone.

Speaking about Re-Invite… I can’t believe that it may be the case. As I already wrote my ATA does not receive any SIP request at all. From ATA point of view Re-Invite begins from normal "Invite" request.

Packet8>> Packet8 Termination Provider>> PSTN>>Provider of PSTN Number (eg. CallCentric) >>SipBroker Server >> Network You're Calling>> Called User's Device
IMHO the feasibility to streamline call from Packet8 via PSTN via SipBroker to User's ATA is doubtful. To do it SipBroker needs knowledge about Packet8 VOIP side – which I guess may NOT come through PSTN link. Do I miss something here?
I compared SIP sessions for two calls (using SipBroker PSTN number which worked): from cell phone and from Packet8 phone.
- in both cases original SIP Invite comes from SipBroker’s IP (64.34.162.221)
- both originally have the same RTP source IP (sipbroker1.telengy.net)
- both re-invited to the same RTP (66.193.176.50)

So? Where we are? Still experimenting… Returning to my original question (forward did not work) – I found out that PAP2T can receive direct IP calls on the same line which already registered with a provider (voipstunt.com) – "Ans Call Without Reg" in PAP2T settings should be changed to "yes". Second condition - UserID should match, but did not in my case. As prove of concept I have coded simple UDP proxy (a program running in PC) – accepting incoming request, replacing UserID and forwarding request to PAP2T – worked like a charm! (As usual there is huge gap between 10-20 lines of code as prove of concept and real program which I have no time to finish). Also I discovered MySipSwitch.com – it seems to employ the same idea…

Once again, thank you guys for the support! VOIP is really exciting area!
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