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2011-04-19
, 09:17
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Posts: 18 |
Thanked: 5 times |
Joined on Apr 2010
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#2
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Shouldnt there be a context=default in sip.conf for the sip peer localphone.com?
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2011-04-19
, 13:41
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Posts: 93 |
Thanked: 13 times |
Joined on Nov 2010
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#3
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Originally Posted by bitwit
Added, but still no change
Shouldnt there be a context=default in sip.conf for the sip peer localphone.com?
anybody?
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2011-04-19
, 14:00
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Posts: 13 |
Thanked: 6 times |
Joined on Apr 2011
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#4
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Originally Posted by rosh
Where is your localphone incoming context? See
Added, but still no change
anybody?
http://help.localphone.com/voip/devi...phone/asterisk
-m
Edit: Please don't use a numeric only username (you are using 1001). It can be any alphanumeric string. You should mix numbers and letters to deter brute force attacks. Suggest you read the voip tech-chat forum on dslreports.
Last edited by voip_wire; 2011-04-19 at 14:03.
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2011-04-19
, 15:10
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Posts: 93 |
Thanked: 13 times |
Joined on Nov 2010
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#5
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Originally Posted by voip_wire
Where is your localphone incoming context? See
http://help.localphone.com/voip/devi...phone/asterisk
-m
Edit: Please don't use a numeric only username (you are using 1001). It can be any alphanumeric string. You should mix numbers and letters to deter brute force attacks. Suggest you read the voip tech-chat forum on dslreports.
What should be my sip credential to put the sip client according to localphone code?
http://help.localphone.com/voip/devi...phone/asterisk
Last edited by rosh; 2011-04-19 at 15:12.
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2011-04-19
, 17:31
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Posts: 13 |
Thanked: 6 times |
Joined on Apr 2011
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#6
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Originally Posted by rosh
I don't understand your question. At all.
What should be my sip credential to put the sip client according to localphone code?
http://help.localphone.com/voip/devi...phone/asterisk
If you want to find out your SIP credentials, the link is on the localphone help page, linked above. Your original problem is that you have not defined routing of localphone incoming calls. The default context, one that you posted, does not go anywhere. Don't change the default context. It should be left as you have it, to minimize the chances of abuse. Define another context, localphone-in, you can name it anything you like really, and make sure that all the calls that reach that context get routed to your local user (1001 in your original example).
You do have a localphone incoming number (DID), don't you?
Why do you want to use asterisk in the first place? Don't setup an asterisk server if you are not fully aware of all the security considerations! You definitely don't need it to connect N900 with (most) VOIP service providers.
-m
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2011-04-19
, 19:13
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Posts: 93 |
Thanked: 13 times |
Joined on Nov 2010
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#7
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Originally Posted by voip_wire
could u please post the additional code i need to put and where.
I don't understand your question. At all.
If you want to find out your SIP credentials, the link is on the localphone help page, linked above. Your original problem is that you have not defined routing of localphone incoming calls. The default context, one that you posted, does not go anywhere. Don't change the default context. It should be left as you have it, to minimize the chances of abuse. Define another context, localphone-in, you can name it anything you like really, and make sure that all the calls that reach that context get routed to your local user (1001 in your original example).
You do have a localphone incoming number (DID), don't you?
Why do you want to use asterisk in the first place? Don't setup an asterisk server if you are not fully aware of all the security considerations! You definitely don't need it to connect N900 with (most) VOIP service providers.
-m
Actually, the reason im trying to go with asterisk is that, for whatever reason my phone doesnt work properly with native SIP sip integration. Specially at university. Most probably firewall and porting issues.
I know that OpenVPN would help this. But after researchin many hrs, still i couldnt manage to setup a OpenVPN connection.
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2011-04-20
, 02:12
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Posts: 93 |
Thanked: 13 times |
Joined on Nov 2010
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#8
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Eureka!. I found the issue. It has something wrong with localphone. I tried with voipcheap and works nicely.
Anybody knows how to use stun (or other nat stuff) of voipcheap in asterisk sip.conf to make the connection more stable?
Anybody knows how to use stun (or other nat stuff) of voipcheap in asterisk sip.conf to make the connection more stable?
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2011-04-20
, 15:51
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Posts: 13 |
Thanked: 6 times |
Joined on Apr 2011
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#9
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Originally Posted by rosh
Eureka!. I found the issue. It has something wrong with localphone. I tried with voipcheap and works nicely.
Anybody knows how to use stun (or other nat stuff) of voipcheap in asterisk sip.conf to make the connection more stable?
You have to a bit more specific. What worked?
N900<=>asterisk<=>voipcheap or
N900<=>voipcheap?
You still haven't answered if you have a DID. Localphone calls it an incoming number.
You can set any stun server in N900. Here is how
Code:
1. Go to the relevant sip account in accounts. 2. Advanced Settings 3. Scroll all the way down. 4. Uncheck Auto-detect STUN 5. Set your preferred STUN server (stun.ekiga.net as an example) 6. Leave the STUN port to default (3478)
If you cannot get it to work on the N900, I am not sure why you think it will work with asterisk. At the minimum, you have to have a basic understanding of SIP protocol to troubleshoot NAT related issues with asterisk. But then again YMMV.
-m
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2011-04-20
, 23:19
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Posts: 93 |
Thanked: 13 times |
Joined on Nov 2010
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#10
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Originally Posted by voip_wire
Thanks for your reply.
You have to a bit more specific. What worked?
N900<=>asterisk<=>voipcheap or
N900<=>voipcheap?
You still haven't answered if you have a DID. Localphone calls it an incoming number.
You can set any stun server in N900. Here is how
Stun server setting in asterisk may or may not be available. It depends on the version that you have. AFAIK, it was introduced in asterisk 1.8.Code:1. Go to the relevant sip account in accounts. 2. Advanced Settings 3. Scroll all the way down. 4. Uncheck Auto-detect STUN 5. Set your preferred STUN server (stun.ekiga.net as an example) 6. Leave the STUN port to default (3478)
If you cannot get it to work on the N900, I am not sure why you think it will work with asterisk. At the minimum, you have to have a basic understanding of SIP protocol to troubleshoot NAT related issues with asterisk. But then again YMMV.
-m
Voipcheap -> asterisks on N900 - > SIP client on N900 works.
N900 <> voipcheap worked obviously. However was not receiving calls sometimes, specially at university WiFi may be due to NAT and other routing stuff.
Now it is much much stable than before, hardly miss a call.
SInce my asterisk server and SIP client are local (within N900), i dont really need a STUN to connect to asterisk server. Im looking for something when asterisk connects to voipcheap to make the voipcheap <> asterisk connection more stable.
I dont have a direct DID to my voipcheap. I got a number for IPkall to connect to voip cheap. Few more questions:
1. Still Caller ID doesnt work, May be voipcheap doesnt support it? or have a issue with asterisk code?
(fixed this already
)2. Is there anyway to connect googlevoice directly with asterisks to make calls without waiting for incoming call?
Last edited by rosh; 2011-04-21 at 11:57.
When i get incoming, the phone doesnt ring but i see, the following msg on asterisk CLI
==Using SIP RTP CoS mark 5
Shouldnt be any port forwarding stuff as i tried everything with DMZ on my router.
sip.conf
[general] srvlookup=yes register=>xxx:xxx@proxy.localphone.com [localphone.com] type=peer disallow=all allow=ilbc allow=ulaw allow=gsm dtmfmode=rfc2833 host=localphone.com fromdomain=localphone.com insecure=port,invite qualify=yes fromuser=xxx authuser=xxx username=xxx secret=xxx ;canreinvite=no nat=yes port=5060 qualify=yes canreinvite=no insecure=invite,port transport=udp [1001] type=friend context=phones host=dynamic nat=yes canreinvite=no secret=abc[globals] [general] autofallthrough=yes [default] exten => s,1,Verbose(1|Unrouted call handler) exten => s,n,Answer() exten => s,n,Wait(1) exten => s,n,Playback(tt-weasels) exten => s,n,Hangup() [outgoing_calls] exten => _X.,1,NoOp() exten => _X.,2,Dial(SIP/${EXTEN:1}@localphone.com,20,r) exten => _X.,3,Hangup [phones] include => outgoing_callsLast edited by rosh; 2011-04-19 at 08:26.