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Unread 09-02-2008, 04:08 PM   #1
dfear
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Default Testing SIP-code's - how?

I have sip-code *8604 for the domain sip.dfear.co.uk.

Every time i look at SIPBroker - Provider White Pages there is a red cross next to the entry for sip.dfear.co.uk.

I have the following dns entries:
_iax._udp SRV 0 0 4569 sip.dfear.co.uk.
_sip._udp SRV 0 0 5060 sip.dfear.co.uk.
sip CNAME home.dfear.co.uk.
home CNAME dfear.ath.cx.

are the above entries correct?

How do i test that a sip-code is correctly setup and working?
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Unread 09-02-2008, 09:34 PM   #2
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Default

Quote:
Originally Posted by dfear View Post
I have sip-code *8604 for the domain sip.dfear.co.uk.

Every time i look at SIPBroker - Provider White Pages there is a red cross next to the entry for sip.dfear.co.uk.

I have the following dns entries:
_iax._udp SRV 0 0 4569 sip.dfear.co.uk.
_sip._udp SRV 0 0 5060 sip.dfear.co.uk.
sip CNAME home.dfear.co.uk.
home CNAME dfear.ath.cx.

are the above entries correct?

How do i test that a sip-code is correctly setup and working?
The RED cross next to your Sip Code is likely there because the last time you tested the call did not complete (it has nothing to do with how your domain is set up)

To test, make sure direct SIP URI (not involving SipBroker) calls to user@sip.dfear.co.uk are ringing on your end correctly...

One quick way to do this:
-Go to SIPBroker - EziDial
-Enter a SIP URI valid for your domain in the field
-Press dial
-You should receive a call from the Echo Test Application
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Unread 09-05-2008, 09:06 PM   #3
dfear
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Quote:
Originally Posted by emoci View Post
The RED cross next to your Sip Code is likely there because the last time you tested the call did not complete (it has nothing to do with how your domain is set up)

To test, make sure direct SIP URI (not involving SipBroker) calls to user@sip.dfear.co.uk are ringing on your end correctly...

One quick way to do this:
-Go to SIPBroker - EziDial
-Enter a SIP URI valid for your domain in the field
-Press dial
-You should receive a call from the Echo Test Application
I tested using SIPBroker - EziDial using the sip uri: 271077@sip.dfear.co.uk -- which works just fine!, BUT when i try calling using a SIP-code it does not work.

My sip-code is *8604 and tried calling *8604 271077 using one of the UK pstn access numbers on a regular landline phone.
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Unread 09-07-2008, 09:20 AM   #4
werty98
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Hi,

To test the outbound calls, just call *011188888 from your system.

To test the inbound calls, you may need to register a free account in ANOTHER SIP server that is also connected to SIP Broker and make calls from that account to your system (using *8604xxxx numbers). Another option is to make calls to the nearest SIP Broker's PSTN gateway.

Remember, never try to make call in which the caller and the callee will have the same public IP (e.g. make calls to your own system from your own system). That will cause error.


And here come my question . Could you tell me how to apply for such a sip-code like yours (*8604)? I've tried to register a new member account, but it allow me to get just a SIP Broker alias (for normal SIP user) but not a sip-code (for SIP server).
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Unread 09-07-2008, 03:57 PM   #5
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Quote:
Originally Posted by werty98 View Post
Hi,

To test the outbound calls, just call *011188888 from your system.

To test the inbound calls, you may need to register a free account in ANOTHER SIP server that is also connected to SIP Broker and make calls from that account to your system (using *8604xxxx numbers). Another option is to make calls to the nearest SIP Broker's PSTN gateway.

Remember, never try to make call in which the caller and the callee will have the same public IP (e.g. make calls to your own system from your own system). That will cause error.
Outbound calls work fine. Incoming calls work fine when calling a sip-uri, but NOT when calling a sip-code. Calling 2001@sip.dfear.co.uk works, but calling *86042001 (which goes to the same extension as the sip-uri) doesn't.

I tried two UK SIP Broker PSTN gateway's and tried calling *86042001 and neither made the (soft)phone ring.

Quote:
Originally Posted by werty98 View Post
And here come my question . Could you tell me how to apply for such a sip-code like yours (*8604)? I've tried to register a new member account, but it allow me to get just a SIP Broker alias (for normal SIP user) but not a sip-code (for SIP server).

Just go to SIPBroker - Provider White Pages and fill in the box labeled SIP Proxy under Add Proxy. Enter in the box the domain name for your server which runs the sip proxy / asterisk server.
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Unread 09-08-2008, 12:25 AM   #6
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Thanks, I've got one *9306 . Everything works fine in the first try of my SIP-code. I've tested to call my PBX from PSTN gateways as well.

Your problem seems to be due to the configuration in your own SIP server. You have not ever been successful to make an inbound call to *8604xxxx, have you? It does not work because you SIP server does not accept calls from SIP Broker. Follows are what I did for my Asterisk PBX, not sure if you need them:

In sip.conf:
[sipbroker]
type=peer
context=sipbroker_inbound
fromuser=421
;fromdomain=www.werty98.homelinux.com
fromdomain=60.240.116.59
host=sipbroker.com
port=5060
incominglimit=3
insecure=invite
nat=yes
canreinvite=no

In extensions.conf, you need to define the context [sipbroker_inbound]. For example, if you want to enable calling *8604271077 from SIP Broker's partners, the extension 271077 must exist in [sipbroker_inbound] in your dialplan.

Cheers,

Werty

Last edited by werty98; 09-08-2008 at 12:41 AM.
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Unread 09-08-2008, 12:36 AM   #7
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Quote:
Originally Posted by werty98 View Post
Thanks, I've got one *9306 . Everything works fine in the first try of my SIP-code. I've tested to call my PBX from PSTN gateways as well.

Your problem seems to be due to the configuration in your own SIP server. You have not ever been successful to make an inbound call to *8604xxxx, have you? It does not work because you SIP server does not accept calls from SIP Broker. If you also use Asterisk, I may tell you what I have done on my Asterisk PBX.
I use an Asterisk based pbx and would like to know how you configured yours to accept incoming calls from sipbroker. Please send me a PM if you don't want to post the details here.

thanks
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Unread 09-08-2008, 12:43 AM   #8
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Quote:
Originally Posted by dfear View Post
I use an Asterisk based pbx and would like to know how you configured yours to accept incoming calls from sipbroker. Please send me a PM if you don't want to post the details here.

thanks
I've just edited my previous post, the configuration is there. Cheers.
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Unread 09-09-2008, 02:05 AM   #9
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Hi,

I've recently found that the above configuration may not work well for inbound calls from some PSTN gateways. The calls are connected without audio transmitted. The problem seems to be due to canreinvite=no.

I am trying to figure out a solution.
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