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DogFace056
join:2005-12-09
Cary, NC

DogFace056

Member

[Lingo] Calling Lingo users directly via SIP

Unfortunately, sipbroker do not provide a working peering code to reach Lingo subscribers. However, they can be called directly via VoIP/SIP at the following URL: "+1LingoNumber@asdcsv.bw.iprimus.net",
where LingoNumber is the Lingo user's 10-digit US number. The same URL may perhaps work for Lingo subscribers in other countries as well, by replacing the 1 with the appropriate country code, although I haven't tested it.

With a PAP2, or equivalent, it is easy to simply assign the URL to a speed dial entry (IP dialling must be enabled).

Alternatively, one may create a dial plan rule to permit calling any arbitrary Lingo user's number. Adding the following example rule entry to the dial plan would do the trick:

<#1,:+>xxxxxxxxxx<:@asdcsv.bw.iprimus.net>

Then simply dial #1 followed by 1 and the 10 digit US phone number.
mazilo
From Mazilo
Premium Member
join:2002-05-30
Lilburn, GA

mazilo

Premium Member

Thank you, DogFace05. It sure will be nice if any asterisk gurus here can provide a dialplan on this hack for asterisk.
slow mo
join:2002-03-19
USA

slow mo to DogFace056

Member

to DogFace056
Interesting, DogFace05! Can it be used as sip uri for sip forwarding from other numbers? If so this will be sooooooo convenient.

DogFace056
join:2005-12-09
Cary, NC

DogFace056

Member

Yes, indeed. In fact, that's how I have my IpKall, FWD and GizmoProject forwarding set up. That way, all my numbers are consolidated through my Lingo line and a single voice mail system. Very practical.

Also, if you're using a DVG-1402s and would like to replace it with a PAP2 (or equivalent), you can configure the PAP2 to work with your Lingo service.

You will need to telnet into the DVG-1402s, and issue commands "sip show" and "sip cfg" to get the SIP configuration credentials. Look for parameters "display_name", "auth_username", "auth_password", "server_fqdn" (or "service_domain"--should be the same), "server_port" (should normally be 5060), "outbound_proxy_fqdn", and "outbound_proxy_port" (should normally be 5060).

Enter those settings into the applicable fields of your PAP2 (under the Line1/2 tab):
Proxy: server_fqdn[:server_port]
Use Outbound Proxy: yes
Outbound Proxy: outbound_proxy_fqdn[:outbound_proxy_port]
Display Name: display_name
User ID: auth_username
Password: auth_password
Use Auth ID: yes
Auth ID: auth_username
You may also want to erase the "SIP User Agent Name" entry under the SIP tab, so that the PAP2 doesn't give away that it's not a DVG-1402s.

Note that doing this may violate the Lingo TOS and lead to service cancellation. If you do this, it's at your own risk.
slow mo
join:2002-03-19
USA

4 edits

slow mo to DogFace056

Member

to DogFace056
Thanks, DogFace05.

I am using SPA1001 (from pap2v1), sip agent name erased, thanks. Lingo is registering and working.

I tried +1aaaeeennnn@asdcsv.bw.iprimus.net with ipKall and it didn't work. It keeps ringing on my cell but my Lingo phone doesn't ring. When I forward ipKall to FWD it works. Any suggestions?

Could it be something else other than asdcsv? Where can I find it? I noticed outbound proxy's 1st 2 letters are state dependent.

ipKall settings
SIP Phone Number: +1aaaeeennnn
SIP Proxy: asdcsv.bw.iprimus.net

DogFace056
join:2005-12-09
Cary, NC

DogFace056

Member

Try the server_fqdn (or service_domain) parameter found in your DVG-1402's configuration. It could be that this server varies depending on service location, just like the outbound proxy.
slow mo
join:2002-03-19
USA

1 edit

slow mo to DogFace056

Member

to DogFace056
Well, I changed to the server domain and it rings but only one way voice. Lingo phone can hear but cell phone hears nothing. After a few seconds my Lingo failover # rings and I can have 2 way voice. Strange, it's like Lingo network thinks my Lingo phone is not registering.

Back to FWD Luckily I am using SPA1001.

DogFace056
join:2005-12-09
Cary, NC

DogFace056

Member

This may be a NAT problem. Have you tried forwarding the RTP ports (default 16384-16482) to your SPA1001?
slow mo
join:2002-03-19
USA

slow mo to DogFace056

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to DogFace056
My router (DI-524) doesn't have RTP port forwarding. It has TCP, UDP, ICMP and *. Don't know what * means, maybe all?

Why when I use FWD it works without forwarding any ports?
caseydoug
join:2001-08-14
Seattle, WA

caseydoug

Member

said by slow mo:

My router (DI-524) doesn't have RTP port forwarding. It has TCP, UDP, ICMP and *. Don't know what * means, maybe all?

Why when I use FWD it works without forwarding any ports?
RTP runs over UDP, but it doesn't use a standard port range. DogFace05 indicates that your adapter uses UDP 16384-16482. The UTStarcom uses ports starting at 13456, but that can be modified by a setting. As he says, one-way audio is frequently caused by NAT problems, although in my experience, it is the inbound voice that has problems. I did notice that those instructions do not include a Registrar Server address. Registration failure could look like what you have described.

Srerglog
@auler.com

Srerglog

Anon

Is it possible to call other cell phone compainies this way. I can send a text message to number@txt.bell.ca and was wondering if there is a server by which I could call the cell phone.

When someone makes an ata-to-cellphone sip address, is that the same as a sip-to-sip call, so that the cell phone user is not charged for the call?
slow mo
join:2002-03-19
USA

slow mo to caseydoug

Member

to caseydoug
said by caseydoug:

said by slow mo:

My router (DI-524) doesn't have RTP port forwarding. It has TCP, UDP, ICMP and *. Don't know what * means, maybe all?

Why when I use FWD it works without forwarding any ports?
RTP runs over UDP, but it doesn't use a standard port range. DogFace05 indicates that your adapter uses UDP 16384-16482. The UTStarcom uses ports starting at 13456, but that can be modified by a setting. As he says, one-way audio is frequently caused by NAT problems, although in my experience, it is the inbound voice that has problems. I did notice that those instructions do not include a Registrar Server address. Registration failure could look like what you have described.
My Lingo account registers and works without any problems. I will do UDP port forwarding when I get home and try it again.

The strange thing is the sip call connects with one way voice (inbound) and after a few seconds my Lingo failover # rings. I can have 2 way conversation on the failover #.

DogFace056
join:2005-12-09
Cary, NC

DogFace056

Member

Looks like there are some problems when forwarding from other SIP providers. It works flawlessly for me when calling directly from another ATA, or from a GP softphone, but forwarding from GP has the one-way speech problem you indicated. And e164.org will ring the phone, but still not pass the test to allow setting up that URI. Could it be that there's a compatibility problem with Lingo's RTP implementation? Hmmm...
maverick215
join:2002-10-03

maverick215 to DogFace056

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to DogFace056
ok... I'll bite. How do you telnet into the DVG-1402?
I know the old pass doesn't seem to work anymore. Do you have the new one?
dony71
join:2005-01-06
San Jose, CA

dony71

Member

do you still have old firmware which unlock DVG-1402S-L from lingo service?