Brekeke Forum Index » Brekeke PBX Forum

Post new topic   Reply to topic
problem with incoming call
Author Message
amr_elsawi
Brekeke Talented


Joined: 19 Nov 2006
Posts: 54
Location: Kuwait

PostPosted: Sat Nov 10, 2007 11:02 am    Post subject: problem with incoming call Reply with quote

Dear All

- I have main Brekeke PBX, and I need to make another PBX which will be registered to the main PBX sip server.

- I can call any UA which connect to the main PBX from the internal PBX.

- when I call from UA( which connected to main PBX), to the internal PBX, and want to call any internal extension, I hear this meesage " the person you want to reach is unavailable, to leave a message , please wait for the tone".

I need to ask why this happened with me, please give me a solution for this.



Thank you

_________________
Amr
Back to top
View user's profile Send e-mail Yahoo Messenger MSN Messenger
pjchacon
Brekeke Addict


Joined: 27 Mar 2005
Posts: 30
Location: Hoboken, NJ

PostPosted: Sun Nov 11, 2007 11:39 am    Post subject: Reply with quote

Hi Amr,

Are both your PBX's on the same network and or VLAN?

It would be helpful to better understand you current topology.

Can you select one of the patters from:
http://www.brekeke-sip.com/bbs/network/networkpatterns.html

It sounds like you have Pattern 7, but please confirm.

Quote:
" the person you want to reach is unavailable, to leave a message , please wait for the tone".


This message is typical of the far end UA not being registered or that the user's Ringer time (sec) setting are too short.

The default setting is 90sec. I tweak mine to be anywhere from 20 to 35 secs.

You can find this setting under the User, Call forwarding settings,
Ringer time (sec).

I hope this helps.

Regards,
Pablo

_________________
FWD# 513461
(1) Cisco 7970, (2) Cisco 7960, (1) Cisco SPA941, (3) BudgeTone-100, ZyXEL P2000W WiFi phone, X-Lite, SJPhone, Cisco ATA 186, HT-488, OnDO PBX & SIP Server, Vonage
Back to top
View user's profile Send e-mail Yahoo Messenger
amr_elsawi
Brekeke Talented


Joined: 19 Nov 2006
Posts: 54
Location: Kuwait

PostPosted: Mon Nov 12, 2007 4:26 am    Post subject: Reply with quote

hi pablo

The problem I think does not in ring time( I was cange it as you told but not solve the problem), but I think its from that the PBX B registered to PBX A, and not registered to itself Sip server, So that there is no call routing between the PBX B and its Sip server.


Do you think that this is the problem.

Thanks
Amr

_________________
Amr
Back to top
View user's profile Send e-mail Yahoo Messenger MSN Messenger
voipwell.com
Partner PBX


Joined: 20 Sep 2005
Posts: 528
Location: Tannersville, Pennsylvania

PostPosted: Mon Nov 12, 2007 12:29 pm    Post subject: Reply with quote

Hello,

You didn't mention what version you are on but let me guess it's 2.0.7.2. If it is and you are using xp or windows 2003 then i would suggest the newest beta released on 10/26 which is the only way i was able to fix the same problem after many hours of changing every setting imagineable.

Nick
Back to top
View user's profile
amr_elsawi
Brekeke Talented


Joined: 19 Nov 2006
Posts: 54
Location: Kuwait

PostPosted: Tue Nov 13, 2007 3:45 am    Post subject: Reply with quote

Dear Nick

I'm using PBX 2.0.7.2 on windows 2003 server.

Now I was make an upgrade for the pbx with the 2.1 Beta version, but the same problem appears.

Do you try before to connect Brekeke PBX to an ITSP, and in the same time connect it to a PSTN GW to keep the extensions in touch with the local phones, and keep the ITSP for international calls???

I think the problem comes from that the PBX not registered to the Sip Server which bundle with the PBX, because when I configure the sip address in the PBX configuration, at this time I can make calls in between extensions, but not to the ITSP because the PBX does not registered to him at this time.

Do you agree with me in this point?

Thank you
Best regards

_________________
Amr
Back to top
View user's profile Send e-mail Yahoo Messenger MSN Messenger
peng
Brekeke Guru


Joined: 20 Jul 2005
Posts: 110

PostPosted: Tue Nov 13, 2007 5:01 pm    Post subject: Reply with quote

Hi amr_elsawi,

Can you see the line "&net.registrar.thru.catchhere=false"
at your dialplan?
Can you try leaving it if you have?
Back to top
View user's profile
amr_elsawi
Brekeke Talented


Joined: 19 Nov 2006
Posts: 54
Location: Kuwait

PostPosted: Thu Nov 15, 2007 6:07 am    Post subject: Reply with quote

Dear Peng

what you want to do with this dial plan?

I make it "&net.registrar.thru.catchhere=true", but no change in the action, the same problem appears.

Do you have any idea about the suitable dial plan for this situation.

Thank you
Best Regards

_________________
Amr
Back to top
View user's profile Send e-mail Yahoo Messenger MSN Messenger
amr_elsawi
Brekeke Talented


Joined: 19 Nov 2006
Posts: 54
Location: Kuwait

PostPosted: Thu Nov 15, 2007 6:41 am    Post subject: Reply with quote

Dear Peng

Now I make it as the following:

Matching Patterns
$port = 15060
$localhost = true
$request = ^REGISTER

Deploy pattern:


$action=register
$auth=false
&net.registrar.onlyglobal=false

and its working too much fine.
now I can call the PBX A Extensions, and also I can call any Extension.

Thank you for your and Nick support

_________________
Amr
Back to top
View user's profile Send e-mail Yahoo Messenger MSN Messenger
amr_elsawi
Brekeke Talented


Joined: 19 Nov 2006
Posts: 54
Location: Kuwait

PostPosted: Thu Nov 15, 2007 10:24 am    Post subject: Reply with quote

Dear Peng and Nick


Now I'm facing another problem,
When I want to call from the upper PBX ( PBX A) to PBX B( which registered to PBX A, U found that the call does not go through to the PBX B, and I cant hear the greating of the PBX B.

Note : before the last modification in the Dial plan, this call was go through to the PBX B.

Thank you

_________________
Amr
Back to top
View user's profile Send e-mail Yahoo Messenger MSN Messenger
peng
Brekeke Guru


Joined: 20 Jul 2005
Posts: 110

PostPosted: Thu Nov 15, 2007 1:05 pm    Post subject: Reply with quote

You said PBX B registered to PBX A, right?
Can you see the Registered Client page at SIP Server of PBX A?

Wht are the contact URI and requester?
Back to top
View user's profile
amr_elsawi
Brekeke Talented


Joined: 19 Nov 2006
Posts: 54
Location: Kuwait

PostPosted: Sat Nov 17, 2007 7:51 am    Post subject: Reply with quote

Wht are the contact URI and requester?

the contact URI : sip:33@192.168.1.30:5060
which is the IP of the PBX B,

the Requester:
Requester : 62.150.107.168:1160
which is the IP of the ADSL router which the PBX B is connected to the internet through it.

I put the network interface: 192.168.1.1, 62.150.107.168

Note: I can call from PBX B to PBX A , but not vise versa.
also I can call between PBX B extensions.
also I can call to the local lines through the Gateway which I connect to the PBX, and vise versa.


the dial plan as the following Now:


Matching Patterns
$port = 15060
$localhost = true
$request = ^REGISTER

Deploy pattern:


$action=register
$auth=false
&net.registrar.onlyglobal=false

Thank you for your supporting me.

_________________
Amr
Back to top
View user's profile Send e-mail Yahoo Messenger MSN Messenger
semensato
Brekeke Addict


Joined: 26 Dec 2008
Posts: 46

PostPosted: Sat Feb 07, 2009 4:15 am    Post subject: Reply with quote

I hame a similar setup but with a PBX and a BSS. They are on different networks. I do can call between BSS users. But can not from PBX to BSS and vice-versa. Users are both registered on BSSs (stand alone and PBX's). Any help will be deeply appreciated.
Back to top
View user's profile
james
Brekeke Master Guru


Joined: 10 Dec 2007
Posts: 494

PostPosted: Mon Feb 09, 2009 11:16 am    Post subject: Reply with quote

If a PBX is on a different network, use $addr = <PBX's Address> instead of $lcoalhost.
Back to top
View user's profile
Display posts from previous:   
Post new topic   Reply to topic    Brekeke Forum Index » Brekeke PBX Forum All times are GMT - 7 Hours
Page 1 of 1