I had read that it was possible to dial 800 numbers (i.e., 800-555-1212) through SipBroker with CallCentric. I have the digit map **275*x. which is supposed to get me to SipBroker. If I pick up the phone and dial **275*18005551212 I get a fast busy. Do I have to set something up on my Obi to access SipBroker (besides the digit map)?
If I dial it directly from CallCentric **218005551212 I am told it is a special services number and not reachable from CallCentric. What am I doing wrong?
Tom
Quote from: tome on September 04, 2011, 08:28:21 AM
I had read that it was possible to dial 800 numbers (i.e., 800-555-1212) through SipBroker with CallCentric. I have the digit map **275*x. which is supposed to get me to SipBroker. If I pick up the phone and dial **275*18005551212 I get a fast busy. Do I have to set something up on my Obi to access SipBroker (besides the digit map)?
I believe you have SP1 set as your PrimaryLine.
When you dial **275*18005551212, you're trying to send 75*18005551212 to Callcentric, which isn't allowed by any of your ITSP Profile B DigitMap rules and wouldn't be accepted by Callcentric anyway.
Since the Callcentric format to make a SIP Broker call is **275* + number, you need to dial **2**275*18005551212.
Quote from: tome on September 04, 2011, 08:28:21 AM
If I dial it directly from CallCentric **218005551212 I am told it is a special services number and not reachable from CallCentric. What am I doing wrong?
That's a Callcentric issue and has nothing to do with the OBi.
Quote from: RonR on September 04, 2011, 09:59:17 AM
I believe you have SP1 set as your PrimaryLine.
When you dial **275*18005551212, you're trying to send 75*18005551212 to Callcentric, which isn't allowed by any of your ITSP Profile B DigitMap rules and wouldn't be accepted by Callcentric anyway.
Since the Callcentric format to make a SIP Broker call is **275* + number, you need to dial **2**275*18005551212.
Yes, I have Sp1 as Primary (GV). When I dial your way I get "The number you dialed **2**275... was rejected by the service provider. Reason 503"
Also, what is the point of having **275*x. in the digit map if I am just force dialing the number to SP2 anyway?
Tom
When you dial **2**275*18005551212, what is shown as being sent to SP2 in your Call History?
The OBi uses the concept of having a PrimaryLine where all calls go by default. If you want to use an alternate trunk, you must specify that trunk using **n. It's possible to come up with an SP1 DigitMap that will detect sequences like **275*xx. and automatically redirect them to SP2. For example, many users have US calls go out Google Voice (SP1) by default and international calls go out VoIP (SP2) by default, but you have to configure your PrimaryLine DigitMap in order for this to occur.
Quote from: RonR on September 04, 2011, 10:14:25 AM
When you dial **2**275*18005551212, what is shown as being sent to SP2 in your Call History?
See attatchment.
Quote from: RonR on September 04, 2011, 10:14:25 AM
The OBi uses the concept of having a PrimaryLine where all calls go by default. If you want to use an alternate trunk, you must specify that trunk using **n. It's possible to come up with an SP1 DigitMap that will detect sequences like **275*xx. and automatically redirect them to SP2. For example, many users have US calls go out Google Voice (SP1) by default and international calls go out VoIP (SP2) by default, but you have to configure your PrimaryLine DigitMap in order for this to occur.
Ok, I see. This isn't really critical, but I have heard that GV will not complete some 800 numbers so I was toying with the idea of using CallCentric. If I do get it to work on CallCentric I will play with the SP1 Digit Map.
Tom
Sip Broker isn't currently accessing 18005551212. Try another 800 number such as **2**275*18003472683.
Quote from: RonR on September 04, 2011, 10:40:03 AM
Sip Broker isn't currently accessing 18005551212. Try another 800 number such as **2**275*18003472683.
Cool, I got Discover. Why wouldn't SIP Broker be able to access that (or any) 800 number?
And thanks!
-Tom
Quote from: tome on September 04, 2011, 10:24:37 AM
This isn't really critical, but I have heard that GV will not complete some 800 numbers so I was toying with the idea of using CallCentric.
I have a Voice Gateway set up that handles all my toll-free calling using a SIP URI to IdeaSip (who welcomes toll-free calls from anyone).
Quote from: tome on September 04, 2011, 10:43:12 AM
Why wouldn't SIP Broker be able to access that (or any) 800 number?
The same reason Google Voice won't complete some 800 numbers. ???
Quote from: RonR on September 04, 2011, 10:50:36 AM
I have a Voice Gateway set up that handles all my toll-free calling using a SIP URI to IdeaSip (who welcomes toll-free calls from anyone).
Is this the only setup needed? http://www.obitalk.com/forum/index.php?topic=526.0
Tom
It's on that theme.
I set up Voice Gateway8:
Name : Toll Free
AccessNumber : SP2(proxy.ideasip.com)
DigitMap : (18(00|88|77|66|55)xxxxxxx|<1>8(00|88|77|66|55)xxxxxxx|!18005551212|!8005551212)
Then I added Voice Gateway8 to the PHONE Port DigitMap:
(...|(Mvg8)|(Mpli))
(placement is critical)
and PHONE Port OutboundCallRoute:
...,{(Mvg8):vg8},{(Mpli):pli}
(placement is critical)
Toll Free calls that would otherwise go out the PrimaryLine are plucked off and sent to Voice Gateway8 instead.
BTW, there are several providers that offer free, open to the public, toll-free termination:
tf.callwithus.com
sip.tollfreegateway.com
tollfreetollfree.com
tollfree.sip-happens.com
free.tollfree4free.com
proxy.ideasip.com
I just called 800-555-1212 with my obi through gv and had no problem with any other toll free numbers am I going to have issues with calling toll free numbers?
Quote from: Blacky on September 04, 2011, 12:54:02 PM
I just called 800-555-1212 with my obi through gv and had no problem with any other toll free numbers am I going to have issues with calling toll free numbers?
You won't know until you know :-\ That is essentially why I was investigating this to begin with.
Tom
I know this thread is old, but I am having a problem with the same issue.
I have SP1 set to Callcentric right now, and if I dial **1**275*xxxxxx, once I hit the first * after the 1, I get an immediate error. Why won't it take any more stars? What do I need to change in the digit rules to allow me to enter the **275* or any other 3digit code after ** when I want to use a different provider? I would like to just be able to dial this manually with any 800 number of my choosing, so setting a speed dial would not work. I want it to recognize the **275*, and then accept the next 11 digits at face value, and send all of that to callcentric.
Add the following to the ITSP A DigitMap:
**275*18(00|88|77|66|55|44)xxxxxxx
The number was rejected by the service provider. Reason 404.
I have tried this with various other strings given by RonR on other threads. Something like **275*x. or [x*][x*]. Same error every time. The voice message reads back the full key entry exactly as I dialed it, which should work.
But, if I place the FULL number in a speed dial like SP1(**275*18001234567), it will dial the number and complete the call.
What gives? ???
Please post your full ITSP A DigitMap
Currently from ITSP Profile A
(**275*18(00|88|77|66|55|44)xxxxxxx|1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.)
I hate dialing **n, so I removed all that code from my configuration. Therefore, I can't easily test the exact way you are dialing. I did successfully route a call out a SP trunk by dialing **275*18005551212.
Is SP1 your Phone Port Primary Line?
Do you want to route 800 numbers this way all the time? If yes, you can automatically add prefix "**275*" after dialing a 1-800 number.
Anyway, if SP1 is your Primary Line dial **275*18005551212.
If SP1 is not your Primary Line, add the following to the beginning of the Phone Port DigitMap:
(Msp1)|
Add above after the beginning paren.
Dial **275*18005551212
Dialing **1............. should work, so maybe something else is wrong.
Please post your Phone Port DigitMap
I forgot something if SP1 is not your Primary Line.
Add ,{(Msp1):sp1} to the Phone Port OutboundCallRoute.
It must be to the right of rule {(Mpli):pli}
Phone port digitmap:
([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**1(Msp1)|**2(Msp2)|**9(Mpp)|(Mpli))
Update: speed dial works as stated above. From keypad, just tried **1**275*18005551212, and it worked also. But without the **1, it gives me the 404 error. (the voice tells me I dialed * * 2 7 5 * 1 8 0 0 5 5 5 1 2 1 2) . As far as I know, sp1 is my primary line. Where do I need to check this? Are there multiple places it needs to be entered?
Physical Interfaces -> Phone Port
On my OBi110 Primary Line is the 5th parm from top.
Error code 404 means NOT FOUND.
I think Callcentric or SIPBroker rejected the call.
Did you dial a valid 800 number?
You can check the call history to see if the call was routed out SP1.
Log into the OBi via the Web interface
Click Status
Click Call History
mo832 -> Check your PM
@azrobert,
So I checked my call history. It sent the **275 calls to SP2 google voice, even though I have the digitmap set as noted. Something isn't right. As long as I FORCE sp1, either via **1 or SP1 in speed dial, it now works correctly, where it wouldn't even dial before. I guess the digit map helped there, but it still won't do right without dialing **1. And my primary line is confirmed to be SP1.
Quote from: mo832 on April 07, 2014, 06:38:35 PM
@azrobert,
So I checked my call history. It sent the **275 calls to SP2 google voice, even though I have the digitmap set as noted. Something isn't right. As long as I FORCE sp1, either via **1 or SP1 in speed dial, it now works correctly, where it wouldn't even dial before. I guess the digit map helped there, but it still won't do right without dialing **1. And my primary line is confirmed to be SP1.
**2 is to force SP2 that is why SP2 makes the call.
I think you would need to put something like this in the digitmap.
**275*xxxxxxxxxxx(Msp1)
Do you want to send all 800 numbers via that route? There are better ways to do this.
I am open to any other methods that would be easier. I switched sp1 to callcentric from localphone as I was having problems with the localphone/ipkall service and I'm trying to pinpoint the problems. But at least with localphone, I could just pick up the phone and dial 1-800-xxx-xxxx with no special codes, and the call would go through with no calling credit. To do the same on CC, you must dial **275. So I would like to at least dial these without changing ITSPs and without needing credit. If I can also do a direct dial, that would be great.
I see no reason to reinvent the wheel here so let's go with old faithful. This thread will show you how to set things up so you can dial 1800 numbers the old way for free.
https://www.obitalk.com/forum/index.php?topic=2357.0
Quote from: RonR on January 22, 2012, 03:44:03 PM
This appears to be another 800 number that uses 'Early Media' that's not supported by Google Voice nor a number of VoIP providers.
A not so elegant solution is to call one of the Sip Broker PSTN Access Numbers (http://www.sipbroker.com/sipbroker/action/pstnNumbers) and then dial *18004683510#.
A more elegant solution is to offload all of your toll free calling to a provider that supports 'Early Media'. This requires that you have SP2 configured for SIP:
Physical Interfaces -> PHONE Port -> DigitMap: (...|(Mvg8)|(Mpli))
Physical Interfaces -> PHONE Port -> OutboundCallRoute : ...,{(Mvg8):vg8},{(Mpli):pli}
Voice Gateway8
Name : Toll Free
AccessNumber : SP2(proxy.ideasip.com)
DigitMap : (18(00|88|77|66|55)xxxxxxx|<1>8(00|88|77|66|55)xxxxxxx)
If you don't already have a SIP provider configured on SP2, use the following:
Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Voice Services -> SP2 Service -> AuthUserName : (put anything here)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B
One change to the above we have 1 new 800 area code 1844 that I am aware of.
DigitMap : (18(00|88|77|66|55|
44)xxxxxxx|<1>8(00|88|77|66|55|
44)xxxxxxx)
For reference according to Wikipedia (http://en.wikipedia.org/wiki/List_of_North_American_Numbering_Plan_area_codes#800.E2.80.93899) these are all toll free in the US or reserved for toll free.
822: reserved for future toll-free expansion (see also 800, 833, 844, 855, 866, 877, 888, 880–882, 883-887, and 889 in this list)
**275*18005551212 should not be forced out SP2 because of the **2.
75*18005551212 must match a rule in the ITSP B DigitMap to be forced out SP2.
(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.) is the default DigitMap for an OBi110.
Even xx. will not match the above string because of the "*"
What ever the reason it's being forced out SP2, you can fix it by placing the following at the beginning of the Phone Port OutboundCallRoute:
{(Msp1):sp1},
Anyway, giqcass' post of RonR's solution is better.
I use tf.callwithus.com
If your SP2 is not defined you can do the following instead of defining a dummy SIP definition:
sp1(tf.callwithus.com)
Now that you mention it if I remember correctly x should only match a number. I forgot that. There is obviously some information about his setup we are missing.
Do you find tf.callwithus.com superior in any way? I have never had issues with ideasip services but I can't confirm whether they support the new 844 code. I can confirm Sipbroker does support it. There are a lot of options out there for 800 termination and with the few 800 calls I make it's difficult to declare one better then the others.
I checked my profile B, and it is the default and it matches the digitmap posted by azrobert above. I have posted my Profile A strings (and tried the modifications given) and my Phone port string.
I am using sp1 with CC (as stated) and currently using sp2 with GV, so I cannot use any free slots on 2 , correct?
I don't really want to do the global solution just yet, but I would like to be able to just dial an 800 number without the **1. Can you guys help me find what is causing the **2(75) to go to sp2 even though I have a pattern to catch it?
EDIT:
If I do the change that is given above
"What ever the reason it's being forced out SP2, you can fix it by placing the following at the beginning of the Phone Port OutboundCallRoute:
{(Msp1):sp1},"
What will this do to other numbers that I may dial? Would everything be forced to sp1 even when I don't expect it?
If you are using the default Phone Port OutboundCallRoute it won't cause any problems.
With the default you must dial **n prefix to route a call to somewhere other than SP1 (the Primary route).
The following is a better solution and won't affect anything else even if you don't have the default Phone Port OutboundCallRoute.
Remove **275*18(00|88|77|66|55|44)xxxxxxx| from the ITSP A DigitMap.
Do not use {(Msp1):sp1}, in the OutboundCallRoute
Add following at the beginning of the Phone Port OutboundCallRoute:
{(<**275*>18(00|88|77|66|55|44)xxxxxxx):sp1},
Dial 18005551212 and it will be routed out SP1 as **275*18005551212
Quote from: giqcass on April 07, 2014, 11:23:15 PM
Do you find tf.callwithus.com superior in any way?
I use tollfree.future-nine.com and tf.callwithus.com. With Future9 you must prefix the number with "**".
Both of these providers will use the UserID as the outbound CallerID.
proxy.ideasip.com does not send a CallerID.
proxy.tollfreenation.com, tollfreegateway.com, sip.denetron.com and sipbroker.com also send CallerID. I haven't use these recently, so I don't know if they still work.
I need CallerID when calling a few 800 numbers.
so I tried the recent changes to the outbound call route, and dialed 18005551212, and I got the "number is invalid" message, which is from callcentric meaning I don't have credit.
I had already gone back to the original sp1 digitmap as instructed.
This is not going too easy for me :(
Sorry, I think I got up too early this morning.
I fixed the error in my Reply #32
Thanks for the revision. The site was down for a stretch today. It even affected the Obitalk expert page which made changes slow and unreliable. In the meantime, I pretty much removed all the special code and tried the RonR global ideasip solution. It seems to be working, but one weird thing is it filters numbers itself. When you dial 18005551212 it takes you to "Free 411" which is a commercial service, but if you dial the same 800# via sipbroker or GV (and I imagine a POTS company) it answers at ATT toll free information. As long as it does what I want, I will wait on re-trying the Outbound call route solution.
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.
Using a VG is the better method.
When you use CC the call goes thru CC and then to SIPBroker.
When you use a VG the call goes directly to the 800 service provider, eliminating 1 hop.
You will get faster response, but probably only a couple milliseconds.
CallWithUs sends you to the AT&T version.
If you want them change the VG AccessNumber to:
sp1(tf.callwithus.com) or sp2(tf.callwithus.com)
If you want to shorten the dialing do the following.
Add to the Phone Port DigitMap:
411|
Change the Voice Gateway DigitMap to:
(<411:18005551212>|18(00|88|77|66|55|44)xxxxxxx)
Now dial 411 or an 800 number.
I changed my config to the above and it works.
To diagnose the previous routing problem, please post the ITSP B DigitMap and the Phone Port OutboundCallRoute.
Quote from: azrobert on April 08, 2014, 03:04:40 PM
To diagnose the previous routing problem, please post the ITSP B DigitMap and the Phone Port OutboundCallRoute.
Those are posted earlier in this thread. The ITSP B is the default given as in the Obi110 (mine is a 100 but same map). The Outbound call route has had Mvg8 added, but before that it was the default. Current copy is:
{([1-9]x?*(Mpli)):pp},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**9:>(Mpp)):pp},{(Mvg8):vg8},{(Mpli):pli}
Quote from: azrobert on April 08, 2014, 10:26:05 AM
Quote from: giqcass on April 07, 2014, 11:23:15 PM
Do you find tf.callwithus.com superior in any way?
I use tollfree.future-nine.com and tf.callwithus.com. With Future9 you must prefix the number with "**".
Both of these providers will use the UserID as the outbound CallerID.
proxy.ideasip.com does not send a CallerID.
proxy.tollfreenation.com, tollfreegateway.com, sip.denetron.com and sipbroker.com also send CallerID. I haven't use these recently, so I don't know if they still work.
I need CallerID when calling a few 800 numbers.
Caller ID when calling a toll free number can be a time saver. Some of the places I call use my CID to pull up my account so for me that is definitely a plus.
I also do not send my true caller ID to companies I am not familiar with. Since you can't block CID to toll free numbers I currently use a Google account with no GV number attached to it. Then they see that generic Google number in CID that just goes to an answering machine.
I had Pitney Bowes collect my CID without my permission when I called them from work. They started calling me at work and claimed I gave them the number. Their telemarketing tactics border on harassment. The whole thing ended with legal threats and me reporting them to the FCC. I was never a customer. I simply inquired one time about their services. I would like to hope they cleaned up their act.
I think I'll use the info you gave here to create a star code to block my CID to 1800 numbers since my current Google Voice trick will cease to function soon.
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.
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.
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 "|[^*#]@@."
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?
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 @@.
"[^*#]@@." 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 ... :)
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?
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.
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.
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.
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?
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.
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?
I always used 10 digits without dashes.
Try the other combinations yourself and see what happens.
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.
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)
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.