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[Lingo] Calling Lingo users directly via SIPUnfortunately, 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. |
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maziloFrom Mazilo Premium Member join:2002-05-30 Lilburn, GA |
mazilo
Premium Member
2008-Feb-12 12:06 am
Thank you, DogFace05. It sure will be nice if any asterisk gurus here can provide a dialplan on this hack for asterisk. |
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to DogFace056
Interesting, DogFace05! Can it be used as sip uri for sip forwarding from other numbers? If so this will be sooooooo convenient. |
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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. |
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4 edits |
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 |
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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. |
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1 edit |
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. |
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This may be a NAT problem. Have you tried forwarding the RTP ports (default 16384-16482) to your SPA1001? |
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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? |
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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. |
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Srerglog
Anon
2008-Feb-14 6:53 am
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? |
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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 #. |
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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... |
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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? |
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dony71 join:2005-01-06 San Jose, CA |
dony71
Member
2008-May-8 4:39 am
do you still have old firmware which unlock DVG-1402S-L from lingo service? |
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