800 calling through SipBroker (via CallCentric)

Started by tome, September 04, 2011, 08:28:21 AM

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azrobert

How are you using a Star Code to block CID?

I dialed *67188842...... and still got direct access to my account.

You could setup 2 VGs. One with your real CID and the other with no or a phony CID.

giqcass

#41
Currently I have a second Google Voice account.  I did not sign up for Google Voice with that account.  Therefore I don't have a number for that account.  When you make calls with a Gtalk account with no number attached you get a generic outbound CID.  I then set that account up on Simonics and sent them an email to allow no registration access to my account.  I set that up as a voice gateway and remapped my obi to send *67 calls out the Voice gateway.

Here is where I explained the process originally.
http://www.obitalk.com/forum/index.php?topic=5451.0

I got the idea from all of those people that had been complaining about having the incorrect outgoing CID because they set there account up wrong.  For billing reasons it's impossible to block CID to toll free numbers.
Long live our new ObiLords!

azrobert

Quote from: mo832 on April 08, 2014, 01:00:55 PM
Now, disregarding all of the special alternatives we have tried, can you help me figure out why my setup was still requiring the **1 even though the codes seemed to take care of it automatically via Primary Line? If I have a logic error somewhere, I would like to find it now before it becomes an issue, if ever.

I took another look at this and found the problem.

ITSP B DigitMap
(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.)

Phone Port OutboundCallRoute
{([1-9]x?*(Mpli)):pp},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**9:>(Mpp)):pp},{(Mvg8):vg8},{(Mpli):pli}

First, the OutboundCallRoute processes left to right. {(Mpli):pli} is the rule that routes calls to the Primary Line (SP1)

[^*#]@@. is the rule causing the problem.
This rule will match a dialed number where the 1st character is NOT an * or # followed by 1 or more characters.

**275*18005551212 will match {(<**2:>(Msp2)):sp2} and be routed to SP2.

**2 will match <**2:>
and
75*18005551212 will match (Msp2)

You can fix the routing problem by removing "|[^*#]@@."

mo832

Thank you azrobert, you are da man!

Regarding that last string that should be removed, will that eliminate any functionality?

I remember another string suggested by RonR that is a better choice. It was supposed to be used for SIP dialing "anything@anything". But how can you dial characters from a phone?

azrobert

I don't know what that rule is for. Removing it shouldn't cause problems.

I removed all rules I don't use.
My SP1 DigitMap looks like this:
(1xxxxxxxxxx|<1>xxxxxxxxxx|<1480>xxxxxxx)

I removed (Mipd). It allows you to dial IP addresses and I don't need it.
I also removed "xx.". It checks for any number of digits.
This can slow down dialing and should be removed if not needed.
When you dial 18005551212 the OBi will wait a couple seconds before routing the dialed number to a trunk because of "xx.". It doesn't know if you will dial additional digits, so it waits.
The call will be routed immediately without "xx.".
I even removed "011xx.". We only call about 5 international numbers and I setup speed dials for them.

RonR did some fancy coding. For an example he would build a URI in the Phone Port DigitMap like "userdi@somewhere.dyndsn.com" and then check for it in the OutboundallRoute with @@.

ianobi

"[^*#]@@." is the original Obihai rule to allow for sip uri calling - "anything@anywhere.com"

"@@." allows any number of almost any alphanumeric characters. As pointed out earlier in this thread, it matches any digits as well as the intended use of sip uri addresses.

RonR suggested using "[^*#]@@.'@'@@." This rule needs a literal @ somewhere in the string before it will be matched, so now it can only match sip uri addresses and not digits that do not contain a "@".

I agree with azrobert, best advice is to drop any rules not being used.

Just tying up the loose ends here ...   :)

mo832

I got error messages on my phone when trying to dial 8xx numbers today using

proxy.ideasip.com

I went in and changed it to sip.tollfreegateway.com and 8xx numbers again went though.

1. Anyone else having the same problem with ideasip?

2. With both ideasip (for months) and now with tollfreegateway I am being told that it sends a random CallerID name/number. It is different every time, but it isn't mine and it isn't a standard number like the Google switchboard or a communications co. Anyone else notice this also? If so, which TF gateways are you SURE send the proper CID?

azrobert

Here is my list of tollfree providers:
tf.arctele.com
tf.callwithus.com
sip.tollfreegateway.com
tollfree.alcazarnetworks.com
sip.denetron.com
sip.broker.com *1800...
sip.denetron.com

My previous testing showed they all passed CID correctly. Testing ideasip showed it did not pass callerid. I just tried one call with tollfreegateway and it worked. The userid used when defining the tollfree trunk will be used as the outbound callerid.

I was using Callwithus, but recently switched to Arctele. I made an 844 call that failed. Arctele and Sipbroker were the only providers that worked with the 844 number.


mo832

Did ideasip work for you to complete the call? It failed for me multiple times today.

Where/ How  do you set the outbound caller ID? And can you make it anything you want? Both name and number? I don't see anything that looks like it applies.

azrobert

I just now tried a call with Ideasip and the call failed with error "407 Proxy Authentication Required". It looks like they now require you to setup an account.

The UserID for the trunk will be used as the outbound CallerID. For a SP trunk it's AuthUserName and for a VG it's AuthUserID. You can't set the name.

mo832

I am using VG 8 for outbound 800 calling. Up until now my AuthUserID and AuthPassword have been blank. I have never populated these fields.

Do I need a password? If so, what do I put here?
Does AuthUserID have to be a number? Can it be any number you wish?

azrobert

Password is not required.
You can set the UserID to anything, but I don't know if the tollfree providers will pass a non-numeric UserID as the CallerID.

I have an account that I can access via 800 number. This account has a feature where I can store several phone numbers. If I call with one of the stored numbers I get immediately connected to the account, otherwise I have to enter a long PIN. This is also how I test if a tollfree provider is passing callerid.

mo832

So what you are saying is if I want to put for example 212-333-4444 as my UserID, it will take that and it will pass that exact number to the 800# incoming phone?

Do I leave out the dashes? Can I put a 5 or 6 digit number or does it have to be a certain length?

As it has been blank all this time, is that why it always throws a random non-related-to-me  "real" number to the recipient?

azrobert

I always used 10 digits without dashes.
Try the other combinations yourself and see what happens.

mo832

Quote from: azrobert on September 12, 2014, 04:21:17 PM
Here is my list of tollfree providers:
[removed]
sip.broker.com *1800...
[removed]

My previous testing showed they all passed CID correctly. Testing ideasip showed it did not pass callerid. I just tried one call with tollfreegateway and it worked. The userid used when defining the tollfree trunk will be used as the outbound callerid.

I was using Callwithus, but recently switched to Arctele. I made an 844 call that failed. Arctele and Sipbroker were the only providers that worked with the 844 number.



So in the list, what does the "*800..." mean for sipbroker? I did not understand that part.

Also, I'm here to report that using tollfreegateway, it DOES pass CID info, but not always. I called a friend with inbound 866 and he tells me that it shows my custom input number half the time, and another random auto-filled number the other half. Plus, many times when calling this way, he told me that my phone sounded like crap...like far away and breaking up. It was corrected on a redial. So I'm ready to try another one.

azrobert

You must send the call in this format:
*18005551212

This DigitMap will auto prefix the dialed number with a star:
(<*>18(00|88|77|66|55|44)xxxxxxx)

ArcTele

Greeting from ArcTele.

I noticed in this thread some issues with caller ID on toll free calls.
Our system is set to pass valid caller ID to our providers.
With all of our testing, all of the providers we use pass it.

If you are having issues, please concat support at arctele.com with your caller ID number, destination number and the date/time. Unfortunately, due to the massive numbers of calls that we process, we only hold call trace details for 48 hours. If you can get us this info in time, we will be glad to look up the call and help get the issue resolved.

Regards,

Zac Amsler, CTO
ArcTele Communications, Inc.