2 Google Voice accounts set to SP1 and SP2.
How do I get a dial tone [in order to call out] on SP2 while on a call on SP1? I will be using 2 different phones.
An OBi100/110 supports only single-line telephones connected to the PHONE Port. Having multiple single-line telephones connected to the PHONE Port does not allow separate simultaneous calls. You can use the hook-flash to place a call on hold while you place a second call (and optionally conference the two calls together).
It is possible to connect additional telephones to the OBi using outboard ATA's, each of which can place calls independent of the telephone connected to the OBi PHONE Port.
Please explain 2 calls per session in the following quote:
http://www.obitalk.com/forum/index.php?action=printpage;topic=190.0
New firmware update (1892) added "Each Google Voice service supports 2 call sessions".
I am not phone tech knowledgeable so you may need to dumby down your explanation.
What does this feature provide? I had interpreted this to mean two calls per session [at one time/simultaneously].
How do I hook-flash? Can this be done with a standard consumer phone? I tried using the flash button on my phone, but I did not receive a [second] dial tone.
Do I need call waiting for any of these functions? Is call waiting part of Obi or Google voice?
From the other post.
Here is what I did:
setup: sp1- GV (primary line), sp2- other sip service.
testing procedure:
1. Using cell phone1 call into AA and dial out using sp1(GV) to cell phone2 (got connected) .
2. then pickup home phone call cell phone 3. And get a greeting no service available something like that.Session 1 - first call comes into the Obi
Session 2 - call is forwarded to another phone (using the same provider, SP1 set as default for outgoing calls)
At that point placing a call from the phone would require dialing **2 to access the SP2 for the outgoing call because both sessions were already in use on SP1. The "no service is available to complete the call ...... message was received because they dialed as if they were placing the call using the default provider (which already had both sessions active).
Successfully placing the call would have given 3 active sessions, two on SP1 and one on SP2. With 2 sessions used for the first call you can still use the 2 sessions available on SP2, either from the phone port or another call handled by the Obi itself.
The Obi does support 2 calls per session for GV. This means you can have 2 simultaneous calls on the same GV account/number at the same time. The Obi's limitation is to one phone on the phone port, so both calls have to be from the same phone. The calls can be both incoming or outgoing, or one of each. Because only one phone is supported, one of the calls will have to be on hold (unless they are conferenced together).
QuoteHow do I hook-flash? Can this be done with a standard consumer phone? I tried using the flash button on my phone, but I did not receive a [second] dial tone.
Normally the use of the flash button would connect you to another incoming call (using call waiting). Switching between incoming calls works like normal call waiting feature.
A feature of the Obi is to give you access to another service/session while another call is active. To access this feature you hook/flash while on an active call. A dial tone should be presented and you can place a call as normal. All calling rules will apply with disregard to the existing call. This means you can place the second call using a different provider (dial **2, **8 or **9) while leaving the second session available on the first provider.
If I recall correctly, once you have two calls active the call waiting feature is unavailable because hook/flash will be directed to your active calls, not to answer a third incoming call.
QuoteDo I need call waiting for any of these functions? Is call waiting part of Obi or Google voice?
You need call waiting for incoming calls only. Call waiting is a feature of the service provider and/or the device. Google Voice does provide call waiting and the Obi supports it.
Each service provider needs to support multi session, GV supports 2 sessions.
With support for 2 calls if you have one GV account, you can get an incoming call and the Obi can forward or fork that call to another number using the same account. No second service provider needed. If you do have 2 providers set up on your Obi, the Obi can forward or fork an incoming call as described and you will still have another service provider to place or receive calls while the first call is still active, you won't tie up both providers with the same phone call.
Note: Since hook/flash from the phone port is local to the Obi, they initiated a "long hook/flash" feature that will connect the phone port to the PSTN. This allows one to answer call waiting calls from the PSTN provider instead of getting a second dial tone from the Obi.
Quote from: skipro on July 17, 2011, 04:35:34 AM
Please explain 2 calls per session in the following quote:
.
.
.
What does this feature provide? I had interpreted this to mean two calls per session [at one time/simultaneously].
It's actually a number of sessions per provider. If a provider allows 2 sessions, it simply means you can have two calls in progress through that provider at any particular time. Google Voice is limited to two sessions. Some VoIP providers allow large numbers of sessions.
Quote from: skipro on July 17, 2011, 04:35:34 AM
How do I hook-flash? Can this be done with a standard consumer phone? I tried using the flash button on my phone, but I did not receive a [second] dial tone.
Hook Flash is initiated by pressing the Flash button on your telephone or momentarily depressing the cradle switch. If you're already on a call, the current call will be placed on hold and you will be given a new dialtone where you can place another call. Hook Flash is also used to answer an incoming call if Call Waiting is enabled. For more information, see the OBi Device Administration Guide.
Quote from: skipro on July 17, 2011, 04:35:34 AM
Do I need call waiting for any of these functions? Is call waiting part of Obi or Google voice?
You do not need Call Waiting from your telephone service provider. If the OBi receives an incoming call while you're already on a call, the OBi provides the Call Waiting indication and allows you take the second call using Hook Flash.
Thanks folks.
I conclude from this that Obi can only handle one active call at a time with 2 GV and a standard landline, regardless of setup or configuration; only 1 call or conversation can be processed at one time with Obi using GV on SP1 & 2 and a landline.
There is no way to carry on 2 calls/conversations simultaneously even with multiple phones connected. The 2 sessions only allow for use of "hold" or conference calling.
Are my conclusions correct?
I am surprised that Obi does not allow at least 2 simultaneous calls.
Each Google Voice account can handle 2 simultaneous calls. VoIP providers may support multiple simultaneous calls.
The PHONE Port can only accomodate 1 call at a time (maybe two if you consider conferencing/call waiting).
But the OBi can also be handling calls other than to the PHONE Port. While you're talking to someone on the PHONE Port, another caller may be using the Auto Attendant to place an outgoing call, or another caller may have been automatically forwarded somewhere, etc.
if you have gv set up on sp1 and a IP phone configured on sp2 would the ip phone ring along with GV message waiting on the phone port?
Could you access the gv line while someone was talking on 1 of the 2 avail gv lines via the ph?
I have Google Voice on SP1 and a VoIP provider on SP2.
I have a telephone connected to the OBi PHONE Port and a telephone connected to a PAP2 which talks to the OBi.
Incoming calls ring both telephones. The one that answers first gets the call.
Both telephones can make separate outgoing calls.
To restate my setup:
1 Obi
2 GV accts for SP1 & 2 on PHONE and a standard land line on LINE. I have several phones connected to PHONE Port, and a standard land line connected to LINE Port.
QuoteEach Google Voice account can handle 2 simultaneous calls. The PHONE Port can only accomodate 1 call at a time (maybe two if you consider conferencing/call waiting).
Are you saying that with only
1 Obi I can make
only 1 active call at a time through my GV acct on PHONE. But if I had
2 OBIs, I could use the same GV acct and
make 2 simultaneous calls 1 through Obi #1 PHONE and the 2nd through Obi #2 PHONE. Is this correct?
QuoteWhile you're talking to someone on the PHONE Port, another caller may be using the Auto Attendant to place an outgoing call,
This is where I get confused.
How does the other caller get to the auto attendant? Can the other caller call thru the same Obi or must it be done on a phone or line not connected to the Obi? If connected to the Obi, do they not need to get a dial tone to get to the auto attendant? How would they get a 2nd dial tone? I cannot get a 2nd dial tone once a call is in place.
Quote from: RonR on July 18, 2011, 03:28:35 PM
I have Google Voice on SP1 and a VoIP provider on SP2.
I have a telephone connected to the OBi PHONE Port and a telephone connected to a PAP2 which talks to the OBi.
Incoming calls ring both telephones. The one that answers first gets the call.
Both telephones can make separate outgoing calls.
RonR:
I'm trying to understand this setup. You have Phone #1 connected to the
PHONE Port of the OBi. You also have a PAP2 Phone Port 1 connected to the
Line Port of the OBi and a Phone #2 connected to
Phone Port 2 of the PAP2. I'm trying to understand why the incoming call ring both Phone #1 (OBi) and Phone #2 (PAP2-Phone Port 2)? Is Line 1 and/or Line 2 of the PAP2 registered to a VoIP provider? I take it that a incoming call to SP2 would also ring "both" phones #1 and Phone #2?
There's no LINE Port connection involved and no use of the Auto Attendant.
For incoming calls:
Voice Gateway5 communicates with the PAP2 via SIP, not the LINE Port. The SP1, SP2, and LINE Port InboundCallRoute's ring both the PHONE Port and the PAP2 with a {ph,vg5(pap2)} rule. The PAP2 is configured to answer calls without registration.
For Outgoing calls:
SP2 is configured for a SIP Provider. The PAP2 is configured with the OBI's IP address and port 5061 for its SipProxy (without registration and make calls without registration).
More can be found here:
http://www.obitalk.com/forum/index.php?topic=718.0
Quote from: skipro on July 18, 2011, 02:20:36 PM
Thanks folks.
I conclude from this that Obi can only handle one active call at a time with 2 GV and a standard landline, regardless of setup or configuration; only 1 call or conversation can be processed at one time with Obi using GV on SP1 & 2 and a landline.
There is no way to carry on 2 calls/conversations simultaneously even with multiple phones connected. The 2 sessions only allow for use of "hold" or conference calling.
Are my conclusions correct?
I am surprised that Obi does not allow at least 2 simultaneous calls.
The Obi can allow as many simultaneous calls as the configured providers allow. But it only allows one active call from the phone port. Other ATAs have more than one phone port, but will likely only allow one active call per port.
You could have an active call using the phone port and place another call using the Obion PC or Smartphone app, or another Obi using the Obitalk Service.
skipro,
You can make a call from the phone port of Obi 1 and then call into Obi 1 from Obi 2 and access the AA to place a second call through the same GV account.
You can set up one GV account on 2 Obi's, but you can't use them both at the same time. The last Obi to connect to the server will be the one that signs into GV.
Quote
This is where I get confused.
How does the other caller get to the auto attendant? Can the other caller call thru the same Obi or must it be done on a phone or line not connected to the Obi? If connected to the Obi, do they not need to get a dial tone to get to the auto attendant? How would they get a 2nd dial tone? I cannot get a 2nd dial tone once a call is in place.
must it be done on a phone or line not connected to the Obi?yes, The AA is accessed using caller id of an
incoming call. This is what the Circle of Trust is for.
Ron could i use this process to connect my LINKSYS SPA9642NA Ip Phone and Cisco 7960?
Quote from: RonR on July 18, 2011, 04:21:16 PM
There's no LINE Port connection involved and no use of the Auto Attendant.
For incoming calls:
Voice Gateway5 communicates with the PAP2 via SIP, not the LINE Port. The SP1, SP2, and LINE Port InboundCallRoute's ring both the PHONE Port and the PAP2 with a {ph,vg5(pap2)} rule. The PAP2 is configured to answer calls without registration.
For Outgoing calls:
SP2 is configured for a SIP Provider. The PAP2 is configured with the OBI's IP address and port 5061 for its SipProxy (without registration and make calls without registration).
More can be found here:
http://www.obitalk.com/forum/index.php?topic=718.0
I'm not familiar with those devices, but I would assume so.
so if i put my ipphone on vg1 with a name 7960 would my InBoundCallRoute be:
{7960>911:sp2},{7960>**0:aa},{7960>***:aa2},{7960>(<**1:>(Msp1)):sp1},{7960>(<**2:>(Msp2)):sp2},{7960>(<**3:>(Mvg3)):vg3},{7960>(<**4:>(Mvg4)):vg4},{7960>(<**6:>(Mvg6)):vg6},{7960>(<**7:>(Mvg7)):vg7},{7960>(<**8:>(Mli)):li},{7960>(<**9:>(Mpp)):pp},{7960>(Mtg1):tg1},{ph,vg1(7960)}
Yes, assuming:
1. Your ipphone's username is 7960.
2. You want 911 to go out SP2.
3. Your PrimaryLine is Trunk Goup 1.
Note: The {7960>(<**3:>(Mvg3)):vg3},{7960>(<**4:>(Mvg4)):vg4},{7960>(<**6:>(Mvg6)):vg6},{7960>(<**7:>(Mvg7)):vg7} rules are there because I have SIP providers on VG3, VG4, VG6, and VG7.
so i took out the stuff you have i dont nee ie the other vg's
{7960>911:sp2},{7960>**0:aa},{7960>***:aa2},{7960>(<**1:>(Msp1)):sp1},{7960>(<**2:>(Msp2)):sp2},{7960>(<**3:>(Mvg1)):vg1},{7960>(Mtg1):tg1},{ph,vg1(7960)}
Right now i can call out but the phone isnt ringing?
i should clarify.. i can call out on tg1 but inbound calls dont ring the ipphone.. the ph port is ringing and i can answer on the ph port
Did you put {ph,vg1(7960)} on SP1, SP2, OBiTALK, and LINE Port InboundCallRoute's? This is the rule that rings both the PHONE Port and the ipphone simultaneously on incoming calla.
yeah.. but only on sp1 and sp2 as i dont use obitalk and line...
I think its a registration problem.. How do i unclick X-Registration required on sp2 when my ITSP requires it?
If your ITSP requires registration, you need to leave it enabled.
If your ipphone requires registration to receive calls, that's a problem.
I have this in my ip phone..
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "0"
Proxy Registration is definetly disabled..
Here is my outbound call setup
7960 signals OBI
|Time | 10.10.10.202 |
| | | 10.10.10.200 |
|35.906 | INVITE SDP (g729 g711U g711A telephone-eventRT...pe-101) |SIP From: "3986" <sip:7960@10.10.10.200 To:<sip:**2xxxxxxxxx@10.10.10.200
| |(50153) ------------------> (5061) |
|35.911 | 100 Trying| |SIP Status
| |(5061) <------------------ (5061) |
|35.927 | 180 Ringing |SIP Status
| |(5061) <------------------ (5061) |
|39.194 | 200 OK SDP (g729 telephone-eventRTPType-101) |SIP Status
| |(5061) <------------------ (5061) |
|39.312 | RTP (g729) |RTP Num packets:336 Duration:6.700s SSRC:0x353006EB
| |(19110) ------------------> (16810) |
|39.360 | ACK | |SIP Request
| |(50153) ------------------> (5061) |
|39.363 | RTP (g711U) |RTP Num packets:1 Duration:0.000s SSRC:0x1103644B
| |(19110) <------------------ (16810) |
|39.379 | RTP (g729) |RTP Num packets:336 Duration:6.699s SSRC:0x1103644B
| |(19110) <------------------ (16810) |
|46.083 | BYE | |SIP Request
| |(50153) ------------------> (5061) |
|46.086 | 200 OK | |SIP Status
| |(5061) <------------------ (5061) |
OBI SETS UP 7960s call to callcentric..
|Time | 10.10.10.200 |
| | | xxxxxxxxxxxx|
|35.926 | INVITE SDP (g729 g711U g711A telephone-eventRT...pe-101 G726-) |SIP From: <sip:xxxxxxxxx@callcentric.com To:<sip:xxxxxxxxxx@callcentric.com
| |(5061) ------------------> (5060) |
|36.009 | 407 Proxy Authentication Required |SIP Status
| |(5061) <------------------ (5060) |
|36.012 | ACK | |SIP Request
| |(5061) ------------------> (5060) |
|36.015 | INVITE SDP (g729 g711U g711A telephone-eventRT...pe-101 G726-) |SIP From: <sip:xxxxxxxxx@callcentric.com To:<sip:xxxxxxx@callcentric.com
| |(5061) ------------------> (5060) |
|36.099 | 100 Trying| |SIP Status
| |(5061) <------------------ (5060) |
|36.166 | 100 Trying| |SIP Status
| |(5061) <------------------ (5060) |
|38.182 | 183 Session Progress SDP (g729 telephone-event...Type-101) |SIP Status
| |(5061) <------------------ (5060) |
|38.184 | RTP (g729) |RTP Num packets:400 Duration:7.961s SSRC:0x83466914
| |(16812) <------------------ (52580) |
|38.657 | 183 Session Progress SDP (g729 telephone-event...Type-101) |SIP Status
| |(5061) <------------------ (5060) |
|39.187 | 200 OK SDP (g729 telephone-eventRTPType-101) |SIP Status
| |(5061) <------------------ (5060) |
|39.192 | ACK | |SIP Request
| |(5061) ------------------> (5060) |
|39.313 | RTP (g729) |RTP Num packets:336 Duration:6.700s SSRC:0x1DE8E623
| |(16812) ------------------> (52580) |
|46.088 | BYE | |SIP Request
| |(5061) ------------------> (5060) |
|46.163 | 200 OK | |SIP Status
| |(5061) <------------------ (5060) |
ron is this the correct sytax for the VG config?
Number : SP2(192.168.1.145:5062)
exactly like this SP2(192.168.1.145:5062)
not just 192.168.1.145:5062
Yes, assuming that your ipphone resides at IP address 192.168.1.145 on port 5062 and that your OBi is configured for SIP on SP2.
I got it working.. my issue was i only had x.x.x.x:port
my 7960 is rocking Gvoice,Callcentric,Li via OBI SP2 on VG1 and VOIPms on seperate lines.
Ron.. Your awesome.
Now for the bad news...
Due to a bug in the OBi firmware, if you ever enable Do Not Disturb (*78) or Call Forwarding All (*72) on the PHONE Port, all processing of all InboundCallRoute rules fails on SP1, SP2, and the LINE Port, breaking all the magic that occurs there. Even though this bug is extremely easy to fix in the OBi firmware, I've been informed by Obihai they have no intention of doing so.
It's a real pity that neat features like this and many many others can't be used along with forwarding incoming calls to your cell phone or temporarily quieting incoming calls, especially when there's no technical reason not to allow it.
what about passing callerid info to the 7960?
Try:
Service Providers -> ITSP Profile x -> SIP -> X_SpoofCallerID : (checked)
that didnt do it.. Does your SPA get calleid?
I actually didn't know as the telephone connected to the PAP2 doesn't display CallerID. I just connected one that does and it appears CallerID isn't making it through.
Passing CallerID through InboundCallRoute rules has been an area where few have had much success. It's not clear whether there are bugs or omissions in this area. Explanations and/or solutions from Obihai have been almost non-existent.
Interesting..
The OBI is def. not passing cID in the sip setup
Here is the call coming into the obi.
|31060.022| INVITE SDP (g711U g729 g711A telephone-eventRT...pe-101) |SIP From: "ALCATEL" <sip:INVITING TN@INVITING IP To:<sip:CALLED TN@ss.callcentric.com
| |(5060) ------------------> (5061) |
|31060.026| 100 Trying| |SIP Status
| |(5060) <------------------ (5061) |
|31060.060| 180 Ringing |SIP Status
| |(5060) <------------------ (5061) |
|31061.091| 200 OK SDP (g729) |SIP Status
| |(5060) <------------------ (5061) |
|31061.249| ACK | |SIP Request
| |(5060) ------------------> (5061) |
|31061.250| RTP (g729) |RTP Num packets:753 Duration:15.036s SSRC:0xB46BE4F2
| |(53036) ------------------> (16830) |
|31061.253| RTP (g711U) |RTP Num packets:1 Duration:0.000s SSRC:0x67AE6007
| |(53036) <------------------ (16830) |
|31061.278| RTP (g729) |RTP Num packets:753 Duration:15.039s SSRC:0x67AE6007
| |(53036) <------------------ (16830) |
|31076.319| BYE | |SIP Request
| |(5060) ------------------> (5061) |
|31076.322| 200 OK | |SIP Status
| |(5060) <------------------ (5061) |
SIP to VG1
|Time | 10.10.10.200 |
| | | 10.10.10.202 |
|31060.058| INVITE SDP (g711U g711A g729 G726-32RTPType-10...726-16RTPTyp) |SIP From: <sip:10.10.10.202 To:<sip:7960@10.10.10.202
| |(5061) ------------------> (5062) |
|31060.265| 100 Trying| |SIP Status
| |(5061) <------------------ (5062) |
|31060.325| 180 Ringing |SIP Status
| |(5061) <------------------ (5062) |
|31061.084| 200 OK SDP (g729) |SIP Status
| |(5061) <------------------ (5062) |
|31061.088| ACK | |SIP Request
| |(5061) ------------------> (5062) |
|31061.251| RTP (g729) |RTP Num packets:753 Duration:15.036s SSRC:0xD83921
| |(16832) ------------------> (22038) |
|31061.277| RTP (g729) |RTP Num packets:756 Duration:15.100s SSRC:0x353006EB
| |(16832) <------------------ (22038) |
|31076.323| BYE | |SIP Request
| |(5061) ------------------> (5062) |
|31076.382| 200 OK | |SIP Status
| |(5061) <------------------ (5062) |
So there are actually 2 calls setup and the messaging on the backside of the OBI seems very minimal. Oh well i am more than happy with my setup. But the nerd in me always wants more.
It's disappointing when things that should work don't and there's not a good reason for it being that way.
hey ron,
any idea why when i dial **0 from my ipphone it rings my phone port and doesnt go to aa?
here is my inbound call statement
{7060>**0:aa}
here is the whole string
{7960>911:li},{7060>**0:aa},{7960>***:aa2},{7960>(<**1:>(Msp1)):sp1},{7960>(<**2:>(Msp2)):sp2},{7960>(<**3:>(Mli)):li},{7960>(<**9:>(Mpp)):pp},{7960>(Mtg1):tg1},{ph,vg1(7960)}
also what would i need to do to call vg1 from the ph? I can dial **9 from the ipphone and get the ph also.
It appears your ipphone is doing something with ** and not passing it to the OBi. Have you checked your ipphone's dial plan?
In the PAP2, I use:
([x*][x*].)
Quote from: bruss on July 21, 2011, 08:12:28 AM
what would i need to do to call vg1 from the ph?
Adding the following will allow the ipphone and PHONE Port to call each other by dialing 0:
{7960>(Mtg1):tg1},
{7960>0:ph},{7960:},{ph,vg1(7960)}
this is my dialplan.xml
<DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>
I'm not familiar with the 7960 or it dial plan syntax. You probably want to set it up to transparently pass all digits and *'s.
in order to pass caller-id info to the IP phone(VGWAYS) you take out the vg1(xyz) statement in the
{ph,vg1(7960),vg2(9642)}
inboundcallrouter on SP2 and replace it with
{ph,SP2(7960@10.10.10.202:5062),SP2(9642@10.10.10.201:5603)}
where 7960 and 9642 are the respective names of my ipphones.
Man what great support I have been getting from OBIHAI... I mean for a 60 dollar product it is just amazing.. I have thousand dollar routers and switches i get less support on..
Thanks for everything.. What a great product
Now, can you get them to fix the OBi so all this doesn't all fall apart if you enable Call Forwarding (*72) or Do Not Disturb (*78)?
If you enable Do Not Distrub (*78) to temporarily quiet the phones, all attempts to make outgoing calls from your 7960 and 9642 will result in a busy signal.
If you enable Call Forwarding (*72) to temporarily forward all incoming calls to your cell phone, all attempts to make outgoing calls from your 7960 and 9642 will result in those calls going to your cell phone.
Quote from: bruss on July 28, 2011, 07:14:21 PM
in order to pass caller-id info to the IP phone(VGWAYS) you take out the vg1(xyz) statement in the
{ph,vg1(7960),vg2(9642)}
inboundcallrouter on SP2 and replace it with
{ph,SP2(7960@10.10.10.202:5062),SP2(9642@10.10.10.201:5603)}
Isn't there a typo in there. 5603 should be 5063?
CID passes because it is an ip phone?
Quote from: bruss on July 28, 2011, 07:14:21 PM
in order to pass caller-id info to the IP phone(VGWAYS) you take out the vg1(xyz) statement in the
{ph,vg1(7960),vg2(9642)}
inboundcallrouter on SP2 and replace it with
{ph,SP2(7960@10.10.10.202:5062),SP2(9642@10.10.10.201:5603)}
where 7960 and 9642 are the respective names of my ipphones.
Am I correct that you also have?:
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerID : (checked)
This method does work, although it souldn't be necessary. I hope Obihai intends to fix Voice Gateway operation as it's much cleaner to have the IP addresses in one place rather than four.
Quote from: QBZappy on July 28, 2011, 07:54:35 PM
CID passes because it is an ip phone?
No, this method works because there's a problem with CallerID handling through Voice Gateways.
Quote from: RonR on July 28, 2011, 09:01:27 PM
Quote from: QBZappy on July 28, 2011, 07:54:35 PM
CID passes because it is an ip phone?
No, this method works because there's a problem with CallerID handling through Voice Gateways.
What problem?
Quote from: QBZappy on July 28, 2011, 09:03:56 PM
Quote from: RonR on July 28, 2011, 09:01:27 PM
Quote from: QBZappy on July 28, 2011, 07:54:35 PM
CID passes because it is an ip phone?
No, this method works because there's a problem with CallerID handling through Voice Gateways.
What problem?
A bug, an omission, a design flaw, only Obihai knows. CallerID is not being passed when a Voice Gateway is used.
RonR,
Be patient until V 1.3 comes out. Until now, CID fix is one of the few carrots that OBi has ever offered the OBi user in advance. They have delivered on promised features/fixes in the past. I don't doubt that they will not fix it. Until then, passing CID is a known problem.
yeah i have a typo.. it should be 5063
And yes X_calledIDSpoof is enabled