problem with incoming call
Moderator: Brekeke Support Team
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- Posts: 54
- Joined: Sun Nov 19, 2006 3:50 am
- Location: Kuwait
problem with incoming call
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
- 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
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/ ... terns.html
It sounds like you have Pattern 7, but please confirm.
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
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/ ... terns.html
It sounds like you have Pattern 7, but please confirm.
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 person you want to reach is unavailable, to leave a message , please wait for the tone".
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
(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
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- Posts: 54
- Joined: Sun Nov 19, 2006 3:50 am
- Location: Kuwait
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
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
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- Joined: Tue Sep 20, 2005 9:10 am
- Location: Tannersville, Pennsylvania
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- Posts: 54
- Joined: Sun Nov 19, 2006 3:50 am
- Location: Kuwait
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
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
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- Posts: 54
- Joined: Sun Nov 19, 2006 3:50 am
- Location: Kuwait
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- Posts: 54
- Joined: Sun Nov 19, 2006 3:50 am
- Location: Kuwait
Dear Peng
Now I make it as the following:
Matching pattern:
$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
Now I make it as the following:
Matching pattern:
$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
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- Posts: 54
- Joined: Sun Nov 19, 2006 3:50 am
- Location: Kuwait
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
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
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- Posts: 54
- Joined: Sun Nov 19, 2006 3:50 am
- Location: Kuwait
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 pattern:
$port=15060
$localhost=true
$request=^REGISTER
Deploy pattern:
$action=register
$auth=false
&net.registrar.onlyglobal=false
Thank you for your supporting me.
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 pattern:
$port=15060
$localhost=true
$request=^REGISTER
Deploy pattern:
$action=register
$auth=false
&net.registrar.onlyglobal=false
Thank you for your supporting me.
Amr