I have successfully set up Google Voice on SP1 and am trying to connect my Grandstream GXP2000 to SP2. I followed the instructions in this article:
http://voxilla.com/2011/02/15/how-to-obify-your-ip-phone-no-asterisk-involved-2623 (http://voxilla.com/2011/02/15/how-to-obify-your-ip-phone-no-asterisk-involved-2623)
However, the details for registering the Cisco phone don't really apply to the Grandstream phone, which is not registering with the OBi100. Can anyone provide guidance that is specific to a Grandstream phone?
After checking out that link I tried it on my GXP2020, I have inbound working, but not outbound yet.
Dave
These are the primary fields (with examples) from the Grandstream GXP2000 account page:
Account Name: (e.g., MyCompany)
SIP Server: (e.g., sip.mycompany.com, or IP address)
Outbound Proxy: (e.g., proxy.myprovider.com, or IP address)
SIP User ID: (the user part of an SIP address)
Authenticate ID: (can be same or different from SIP UserID)
Authenticate Password: (not displayed for security protection)
Name: (optional, e.g., John Doe)
I believe that SIP Server, Outbound Proxy, SIP User ID, and Authenticate ID are the key fields. Would you care to share the inputs that work for you? If you are having partial success, your Grandstream phone must be registering with the OBi100. I would like to get to that point myself.
So far the fields I have changed settings in are
Account Name: GoogleVoice (don't think its important)
SIP Server: (I have tried both leaving this blank and putting in the IP address and port of myOBi110, incoming works both ways, outgoing does not work, but with the OBi110 IP and port I get a "486 Busy Remote party is busy" message, no message with no SIP server setting, but still rings on incoming)
Outbound Proxy: 192.168.1.34:5061 (I have the IP address and port of my OBi110 sp2)
SIP User ID: gxp2020l1
Authenticate ID: (left blank)
Authenticate Password: (left blank)
Name: (left blank)
Also I have the following set:
Use DNS SRV: No
User ID is phone number: No
SIP Registration: No
Unregister On Reboot: No
Support SIP Instance ID No
There doesn't seem to be a setting for "Ans Call Without Reg" like the Cisco phone the writer mentioned so I did nothing with that setting, but I still get my 2020 to ring when I receive a GV call. Both my analog phone plugged into phone port on my OBi110 and my GXP2020 ring and can be answered.
Also I noted that there seemed to be an typo in step 6 of those instructions about setting "X_InboundCallRoute":
Where it says to put in
{spaline5>(911):li},{spaline5>(xxxxxxxxx):pp},spaline5>(xx.):sp}
they forgot a { so it should read:
{spaline5>(911):li},{spaline5>(xxxxxxxxx):pp},{spaline5>(xx.):sp}
and I replaced the spaline5 with my gxp2020l1 ID which I had set on the 2020.
I haven't spent very much time on this yet, and probably won't have time today, but maybe those settings will get you partially working like I am. Let me know if you have anymore questions.
Dave
I discovered this on Google and decided to set up one of my old ATA devices. I never succeeded with getting my PAP2NA to work but my Grandstream HT502 works like a charm.
The only real change I made was to the sp2 X_InboundCallRoute digitmap. I essentially used the same digitmap I have in profile A for SP1. Replace 1234 with your user name. Replace AAA with your area code.
{1234>(911):li},{1234>(xxxxxxxxx):pp},{1234>(*xx|<411:18002464411>S0|[235678]11!|<:1AAA>[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|<:1>[2-9]xx[2-9]xxxxxxS0):sp1}
In the Setup Wizard I also placed ph,sp2(1234) In all three boxes for inbound call route to make both phones ring on any kind of incoming call.
Notes: Don't expect anything to "register". This is a SIP to SIP connection and there is no proxy server in the middle to register with. In your IP phone or ATA, SIP Register should be set to OFF. The same goes for the Obi, uncheck the box for X_RegisterEnable in the sp2 setup. That was not mentioned in the instructions.
In the Grandstream HT502 there is no separate box for server port. Enter it with the IP address for example: 192.168.2.24:5061
You also need to make sure that ITSP B RegistrarServerPort is set at port 5061 if you are following the instructions in the original article. My default was 5060, not 5061 as the article says.
Quote from: M105 on May 14, 2011, 09:02:25 PMI never succeeded with getting my PAP2NA to work but my Grandstream HT502 works like a charm.
M105,
FWIW, I've been wanting to get a PAP2 working with the OBi, so I tore into it tonight and it's working great. All four inbound trunks (sp1, sp2, li, pp) ring the OBi phone and the PAP2 simultaneously and I can do everything from the PAP2 as if it was the OBi phone (for example, Auto Attendant, Device Configuration, SIP Voice Gateway calls, Trunk Group calls, etc.).
SP2 InboundCallRoute:
{pap2>911:li},{pap2>**0:aa},{pap2>***:aa2},{pap2>(<**1:>(Msp1)):sp1},{pap2>(<**2:>(Msp2)):sp2},{pap2>(<**3:>(Mvg3)):vg3},{pap2>(<**4:>(Mvg4)):vg4},{pap2>(<**6:>(Mvg6)):vg6},{pap2>(<**7:>(Mvg7)):vg7},{pap2>(<**8:>(Mli)):li},{pap2>(<**9:>(Mpp)):pp},{pap2>(Mtg1):tg1},{ph,vg5(pap2)}
I already have a SIP VoIP provider on SP2, so calls to the PAP2 get there via Voice Gateway5:
VG5 -> AccessNumber : SP2(192.168.1.145:5062)
Using this approach, there shouldn't be any problem adding another PAP2 or two.
It is incredible what can be done with the Obi Ron. I looked at the voice gateways and thought they could/should be used like that but have not delved into it that deep yet. You should write an "Obi for Dummys" book packed with examples and sell it over the web. (I am serious) A really good tutorial on using some of the advanced features, with working examples, and priced right ($5.00-$10.00/download) would be a winner. You could actually provide some templates in the from of config files...
I think my old PAP2 has some hardware issues which is why I retired it a while back. I never got it to ring the attached phone. I am glad to know that it will work though. There are millions of them floating around out there and they make a good extension to the Obi. In my case I have a shop building 100 yards behind my house with power but no phone line. I have a wifi repeater bridge out there (made from a WAP54g) with one wired ethernet port. I intend to stick this HT502 out there and see how it performs.
I am still struggling with the Grandstream 2000. In the OBi100 Dashboard, under 'Device Configuration - Home' and the section 'Configure Voice Services', 'Service Provider 2' still says 'Not Configured'. When you had the Grandstream 2020 working, what was this setting? Should this not say 'Connected'?
Still would like an answer to the above, but have now got the Grandstream to ring on incoming - I missed a crucial bit of the instructions! (No luck with outgoing). However, when the call comes into the Grandstream, it is on Line 4 (registered with OnSIP) rather than Line 1, which I am using for the OBi100. Line 4 is the only line that is registered to a VOIP account.
When I get some more time, I am going to start from scratch and redo the setup.
Both my service providers say Not Configured in the web portal. They have since I started using the expert config.
Quote from: rmclaren on May 15, 2011, 11:31:33 AM
I am still struggling with the Grandstream 2000. In the OBi100 Dashboard, under 'Device Configuration - Home' and the section 'Configure Voice Services', 'Service Provider 2' still says 'Not Configured'. When you had the Grandstream 2020 working, what was this setting? Should this not say 'Connected'?
Still would like an answer to the above, but have now got the Grandstream to ring on incoming - I missed a crucial bit of the instructions! (No luck with outgoing). However, when the call comes into the Grandstream, it is on Line 4 (registered with OnSIP) rather than Line 1, which I am using for the OBi100. Line 4 is the only line that is registered to a VOIP account.
When I get some more time, I am going to start from scratch and redo the setup.
My SP2 (which have the GXP2020 set up at) shows
Not Configured. You aren't registering to a service with this setup, so it will not show registered. BTW, I haven't been using the OBiTalk Dashboard to configure my OBi110, I use the local web interface on the OBi itself, and both my OBi110 and GXP2020 are configured with static IP addresses. I don't know if that has anything to do with your issues or not. On my GXP2020 my line 1 (GoogleVoice) always shows connected, even when my Internet connection is down. Make sure you have that
SIP Registration option on line 1 set to
NO.
I still haven't gotten it to dial out yet, might be due to not having a SIP account registered, since I have GV on SP1 and my Grandstream GXP2020 on SP2. I think I'll try the Voice Gateway configuration that RonR suggested for my 2020 and configure one of my SIP accounts on SP2, thanks RonR. Being able to answer my GV calls on the Grandstream is nice, but really would like dial out working too. The Grandstream firmware does not support a
Dial Plan setting for each SIP account, I think your GXP2000 has the same firmware as my 2020. I'm at: Program-- 1.2.5.3 Bootloader-- 1.1.6.9. That might be the problem with getting dial out working. If that's the case we would need Grandstream to update the firmware to support dial plans on the SIP accounts, which may not be something they are willing to do for the small number of us who would use it. I have a HT502 also, might try getting that working too just for fun.
But as it sits with the configuration I posted, I get my 2020 ringing on incoming calls, CID comes through just fine, I can answer and talk and everything works great, just haven't gotten dialing out working. Feel free to contact me, or if you really want I could take some screen shots of my 2020's web config page.
Dave
RonR, can you explain this a better for thick skull like me? I have a SIP phone I would like to use and I am a bit confused. First, did you start by added the voice gateway and then add that gateway to an existing route? I only have one service configured, GV, and would like to set up a SIP phone or two and things are just not clear.
Quote from: RonR on May 15, 2011, 12:31:06 AM
Quote from: M105 on May 14, 2011, 09:02:25 PMI never succeeded with getting my PAP2NA to work but my Grandstream HT502 works like a charm.
M105,
FWIW, I've been wanting to get a PAP2 working with the OBi, so I tore into it tonight and it's working great. All four inbound trunks (sp1, sp2, li, pp) ring the OBi phone and the PAP2 simultaneously and I can do everything from the PAP2 as if it was the OBi phone (for example, Auto Attendant, Device Configuration, SIP Voice Gateway calls, Trunk Group calls, etc.).
SP2 InboundCallRoute:
{pap2>911:li},{pap2>**0:aa},{pap2>***:aa2},{pap2>(<**1:>(Msp1)):sp1},{pap2>(<**2:>(Msp2)):sp2},{pap2>(<**3:>(Mvg3)):vg3},{pap2>(<**4:>(Mvg4)):vg4},{pap2>(<**6:>(Mvg6)):vg6},{pap2>(<**7:>(Mvg7)):vg7},{pap2>(<**8:>(Mli)):li},{pap2>(<**9:>(Mpp)):pp},{pap2>(Mtg1):tg1},{ph,vg5(pap2)}
I already have a SIP VoIP provider on SP2, so calls to the PAP2 get there via Voice Gateway5:
VG5 -> AccessNumber : SP2(192.168.1.145:5062)
Using this approach, there shouldn't be any problem adding another PAP2 or two.
ccbadd,
The OBi must have SP1 and/or SP2 configured for SIP. It need not have a working provider. You can set the proxy server on ITSPx to anything (like 127.0.0.1) and disable X_RegisterEnable on SPx.
I already had a working SIP provider configured on SP2/ITSPB.
Line 1 of the PAP2 is configured:
IP Address : 192.168.1.145
SIP Port : 5062
Proxy Server : 192.168.1.140:5061 // SP2 on OBi
Register : No
Make Call Without Registration : Yes
Answer Call Without Registration : Yes
UserID : pap2
Dial Plan : ([x*][x*].)
OBi Voice Gateway5 is set to point to the PAP2:
Name : PAP2
AccessNumber : SP2(192.168.1.145:5062)
Each InboundCallRoute is modified to fork to VG5 in addition to the PHONE Port : {ph,vg5(pap2)}
SP2's InboundCallRoute is a little more complex:
{pap2>911:li},{pap2>**0:aa},{pap2>***:aa2},{pap2>(<**1:>(Msp1)):sp1},{pap2>(<**2:>(Msp2)):sp2},
{pap2>(<**8:>(Mli)):li},{pap2>(<**9:>(Mpp)):pp},{pap2>(Msp1):sp1},{pap2>0:ph},{pap2>(@@.):},
{ph,vg5(pap2)}
This is basically the equivalent of the relevant portions of the PHONE Port OutboundCallRoute. It's used to route input from the PAP2 to the appropriate trunk.
I have Obi110 configured SP1 with GV, SP2 with SIP provider, and SPA1001 configured as per your steps.
It works except
1. when outgoing call from SPA1001, there is one way voice (I can't hear from other side), incoming call to SPA1001 worsk.
2. Caller ID does not show on SPA1001.
Do you have any idea?
Quote from: RonR on May 29, 2011, 10:55:39 AM
ccbadd,
The OBi must have SP1 and/or SP2 configured for SIP. It need not have a working provider. You can set the proxy server on ITSPx to anything (like 127.0.0.1) and disable X_RegisterEnable on SPx.
I already had a working SIP provider configured on SP2/ITSPB.
Line 1 of the PAP2 is configured:
IP Address : 192.168.1.145
SIP Port : 5062
Proxy Server : 192.168.1.140:5061 // SP2 on OBi
Register : No
Make Call Without Registration : Yes
Answer Call Without Registration : Yes
UserID : pap2
Dial Plan : ([x*][x*].)
OBi Voice Gateway5 is set to point to the PAP2:
Name : PAP2
AccessNumber : SP2(192.168.1.145:5062)
Each InboundCallRoute is modified to fork to VG5 in addition to the PHONE Port : {ph,vg5(pap2)}
SP2's InboundCallRoute is a little more complex:
{pap2>911:li},{pap2>**0:aa},{pap2>***:aa2},{pap2>(<**1:>(Msp1)):sp1},{pap2>(<**2:>(Msp2)):sp2},
{pap2>(<**8:>(Mli)):li},{pap2>(<**9:>(Mpp)):pp},{pap2>(Msp1):sp1},{pap2>0:ph},{pap2>(@@.):},
{ph,vg5(pap2)}
This is basically the equivalent of the relevant portions of the PHONE Port OutboundCallRoute. It's used to route input from the PAP2 to the appropriate trunk.
Quote from: Tiger on July 30, 2011, 09:54:15 AM
1. when outgoing call from SPA1001, there is one way voice (I can't hear from other side), incoming call to SPA1001 worsk.
Is the SPA1001 located external to the LAN the OBi is located on or are they both hung off of the same switch?
Quote from: Tiger on July 30, 2011, 09:54:15 AM
2. Caller ID does not show on SPA1001.
CallerID not being passed properly through Voice Gateways is a known problem in the OBi firmware.
Instead of using a Voice Gateway, you should be able to get around this problem by replacing the {ph,vg5(pap2)} rules with {ph,sp2(username@ipaddress:port)}, where username, ipaddress, and port are what you had previously used in the Voice Gateway.
Is the SPA1001 located external to the LAN the OBi is located on or are they both hung off of the same switch?
Answer: Both are the same subnet and behind the same router.
CallerID not being passed properly through Voice Gateways is a known problem in the OBi firmware.
Instead of using a Voice Gateway, you should be able to get around this problem by replacing the {ph,vg5(pap2)} rules with {ph,sp2(username@ipaddress:port)}, where username, ipaddress, and port are what you had previously used in the Voice Gateway.
Answer: caller ID Workss after the changes, thanks
Quote from: Tiger on July 30, 2011, 11:11:20 AM
Answer: Both are the same subnet and behind the same router.
Then I wouldn't have expected one-way audio problems.
Take a look that the NAT Support Parameters in the SPA1001. FWIW, I have them all set to Yes in my PAP2. You might experiment with various combinations to see if they have any effect on your problem.
Quote from: RonR on July 30, 2011, 11:32:28 AM
Quote from: Tiger on July 30, 2011, 11:11:20 AM
Answer: Both are the same subnet and behind the same router.
Then I wouldn't have expected one-way audio problems.
Take a look that the NAT Support Parameters in the SPA1001. FWIW, I have them all set to Yes in my PAP2. You might experiment with various combinations to see if they have any effect on your problem.
Something wrong when call made from SPA1001, the log shows state is ringing even both connected, and one of peer RTP address is 0.0.0.0
Terminal ID SP2 SP2
State ringing connected
Peer Name
Peer Number pap2 1xxxxxxxxxx
Start Time 19:23:36 19:23:36
Duration 00:00:07 00:00:01
Direction Inbound Outbound
Peer RTP Address 0.0.0.0:0 208.x.x.x:63766
Local RTP Address 192.168.1.28:16834 192.168.1.28:16836
RTP Transport UDP UDP
Audio Codec tx=G711U; rx=G711U (bridged) tx=G711U; rx=G711U (bridged)
RTP Packetization (ms) tx=0; rx=20 (bridged) tx=20; rx=0 (bridged)
RTP Packet Count tx=0; rx=92 tx=92; rx=92
RTP Byte Count tx=0; rx=15824 tx=15824; rx=15824
I don't see that probem here.
Issue resovled after set the following in SPA1001
Use Outbound Proxy: yes
Outbound Proxy: <Obi SP2 Ip address>:<port> - the same as Proxy value
Use OB Proxy In Dialog: yes
That must be a quirk in the SPA1001 as I have them both set to No in the PAP2 and no Outbound Proxy specified.
I'm glad you found a solution.
Is there any way to add some security to this?
I wanted to use the Obi110 to call out on Google Voice from an Android Gingerbread phone using the native SIP stack. (not the obion app) The instructions here worked just fine.
However, if I have ports 5060-5061 open to the internet, then anyone can use my obi to call out if they can discover the username.
Is there any way to add authentication? Is the username passed in the open?
QuoteIs there any way to add authentication? Is the username passed in the open?
I don't believe there is any way to make this really secure.
You can get pretty good protection from random attacks by using an obscure port (not 5060 or 5061) and a long pseudo-random user name. However, that won't protect you from someone who can capture your traffic (unlikely over 3G, but certainly possible over open or WEP Wi-Fi).
If you are using this for domestic calling, you could set up the OBi to permit only specific area codes when bridging. If calling internationally, GV rates are not very aggressive -- I'd just use a regular SIP provider instead.
Quote from: Brian01 on October 08, 2011, 04:04:35 PM
Is there any way to add some security to this?
I wanted to use the Obi110 to call out on Google Voice from an Android Gingerbread phone using the native SIP stack. (not the obion app) The instructions here worked just fine.
Brian, can you point to the instructions? All I could find was a thread saying that it was not possible: http://www.obitalk.com/forum/index.php?topic=205.msg930#msg930
OK, I posted my Android config in that thread:
http://www.obitalk.com/forum/index.php?topic=205.msg10762#msg10762
Regarding security, I noticed from above that you can use X_AccessList to limit access by ip address. That's something, anyways.
Thanks everyone for the info.
Thanks Brian, yes I saw them there.
If one has a domain name pointing to one's home network, I suspect one could in fact set it up to access your Obi even when not at home...
something like this:
Obi at fixed IP address 192.168.1.200
Some domain name pointing you your home router (such as from no-ip.com)
Eg myzootyname.com
Home router set to direct incoming traffic on port 5066 to the obi.
then the Obi is essentially at myzootyname.com:5066
now one issue I see is that when we are away from home but on a wifi network, making a phone call will use the home network twice - voice data would go from the phone to the router to the OBi and then out through the router again to the end point. Depending on the speed of the home network connection this may be no problem.
This would mean that one can use it when away from home.
I am also interested in using the native Nexus S stack so that I get fill contact list integration etc. (I did import my contact list into the Obion app but contact changes wont sync up I am sure.
Yeah. You could use the ip address of your home, no need for a domain.
It's also a one line hack of the Android system to make the native sip stack work over cell data in addition to wifi. So you could conceivably use the setup over your cell carrier's 3g or 4g for free cell calling. (wouldn't try it over EDGE)
As you say though, the quality would depend on your home connection, cell tower, etc. I've tried VOIP over T-mobile 3g and it works just fine if you're near a nice strong data node.
What I really want is a better dialer. Right now I use the built-in dialer and it asks if I want to use Google Voice or not...then it asks if I want to use cell or internet...two annoying dialogue boxes. We need a dialer with checkboxes and a better default system.
yes but the IP address of the home router can change and the no-ip DNS entry ensure that it can be found....
I told my android google voice never to dial calls... GV actually uses the cell connection to call into GV and then bridges that to the call you dialed. So for a T-mobile pay as you go customer it achieves nothing.
Quote from: rmclaren on April 21, 2011, 07:31:53 AM
I have successfully set up Google Voice on SP1 and am trying to connect my Grandstream GXP2000 to SP2. I followed the instructions in this article:
http://voxilla.com/2011/02/15/how-to-obify-your-ip-phone-no-asterisk-involved-2623 (http://voxilla.com/2011/02/15/how-to-obify-your-ip-phone-no-asterisk-involved-2623)
However, the details for registering the Cisco phone don't really apply to the Grandstream phone, which is not registering with the OBi100. Can anyone provide guidance that is specific to a Grandstream phone?
Thanks so much for the link!
I used that great guide to set up a few Grandstream GXP2140's with an Obi202. The problem I'm having is transferring calls between phones. All of the phones have EXACT identical settings (except for the IP address).
If a call comes in and I pick it up on one GXP2140 and I want to transfer it to another GXP2140 how can I do it? I tried using Quick IP dialing, but that doesn't support call transfer (just disconnects the originating call).
I asked in Grandstream support chat, but they weren't familiar with this freaky Obi setup "Didn't you just set up your voip providers settings inside the GXP2140??".... "Ummm, no. Because Anveo limits me to two devices logged in, so using my Obi box as the proxy I can have tons of phones all use Anveo..." so they had no idea how call transfer would work. Any ideas??
Another thing that's freaky. I figured if I set up identical settings in each phone, that the "lines" would be shared on each phone (i.e. if I was on a call on phone1 line1, then on phone2, line one would be lit up red, or at least let me barge into it), but nope. If a call comes in on "line 1" and I put it on hold, I can click "line 1" on the other phone, and I get a dialtone, and can make calls no problem!! strange :)
If you have any information on how I can transfer a call from one IP phone to another one logged into the same Obi box please let me know.
-Jamie M.