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[asterisk-users] Incoming Calls


 
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support at drdos.info
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PostPosted: Wed Jan 02, 2008 11:27 am    Post subject: [asterisk-users] Incoming Calls Reply with quote

Paulo Pinheiro wrote:
Quote:

I am having a problem that I would like to verify if someone could
help?I am using bandwith.com as my SIP TRUNK provider. When I place
the phone number in the DID number field ( using Elastix) it gives me
an error message stating the phone number I dialed is not in service.
When I leave the DID number and CLID number blanks

Your best bet would be to ask them. They do have both mailing lists and
forums.

http://sourceforge.net/mail/?group_id=161807
http://sourceforge.net/forum/?group_id=161807

Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
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rakh at dangerclan.net
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PostPosted: Wed Jan 02, 2008 11:50 am    Post subject: [asterisk-users] Incoming Calls Reply with quote

Hi Mr. Paulo,

Could you please explain this situation in a more detailed way to see
how can we help you?

Regards,

Paulo Pinheiro wrote:
Quote:

I am having a problem that I would like to verify if someone could
help...I am using bandwith.com as my SIP TRUNK provider. When I place
the phone number in the DID number field ( using Elastix) it gives me
an error message stating the phone number I dialed is not in service.
When I leave the DID number and CLID number blanks it works fine. I
really need to have the system identifying multiple phone numbers (
multiple trunks ) but I have not been able to do so. Would anyone be
able to help?



Thanks,



Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767



------------------------------------------------------------------------

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Tony at plack.net
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PostPosted: Wed Jan 02, 2008 12:02 pm    Post subject: [asterisk-users] Incoming Calls Reply with quote

Quote:
Hi Paulo,

Make sure your DID number is in the e.164 format, ie, +15551234567.
I had the same issue with bandwidth.com and that fixed the problem.


HTH,

Zaheer

Zaheer is right. Everything from bandwidth is 164 format. So you need the +15551234567 in the dial plan as well as in your sip.conf

Tony Plack
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Paulo.Pinheiro at cent...
Guest





PostPosted: Wed Jan 02, 2008 12:18 pm    Post subject: [asterisk-users] Incoming Calls Reply with quote

Hi Jose, I apologize for the lack of information..I am new to this...Let
me try to be more specific:



I've got Asterisk installed on Linux. I am using Elastix as the front
end to make changes in the system.



Under the Trunk set up these are my setting for the Peer Details:



allow=ulaw&alaw&gsm

auth=plaintext

canreinvite=no

context=from-internal

disallow=all

dtmfmode=inband

fromdomain=xxx.xxx.xxx.xxx (IP address)

host=xxx.xxx.xxx.xxx (IP address)

insecure=very

nat=no

qualify=no

tos=none

type=friend



these are my settings for User Details:



allow=ulaw

canreinvite=no

context=from-sip-external

dtmfmode=rfc2833

host=xxx.xxx.xxx.xxx (IP addres)

nat=no

port=5060

reinvite=no

type=peer



When setting up the income routes if I place the phone number in the DID
Number field, when calling the number I receive a message stating the
phone number is not listed or out of service. When I leave the DID
Number field blank everything works because it does a catch all scenario
but that is not what I am looking for.



I have tried to place the phone number with +1 in front of it and still
does not work. Any way to help?



Thanks much,

Paulo





From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jose P.
Espinal
Sent: Wednesday, January 02, 2008 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls



Hi Mr. Paulo,

Could you please explain this situation in a more detailed way to see
how can we help you?

Regards,

Paulo Pinheiro wrote:

I am having a problem that I would like to verify if someone could
help...I am using bandwith.com as my SIP TRUNK provider. When I place
the phone number in the DID number field ( using Elastix) it gives me an
error message stating the phone number I dialed is not in service. When
I leave the DID number and CLID number blanks it works fine. I really
need to have the system identifying multiple phone numbers ( multiple
trunks ) but I have not been able to do so. Would anyone be able to
help?



Thanks,



Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767






________________________________




_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



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jonnt at taylortelepho...
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PostPosted: Wed Jan 02, 2008 12:19 pm    Post subject: [asterisk-users] Incoming Calls Reply with quote

Paulo,



I am using them also. All call traffic to and from them must be in e164 format. So your calls have to look like this, +15615551212 or +011 for international. They do not let you set the caller name, but will let you set calling number and that also needs to be e164 format.



Jonn



_____

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paulo Pinheiro
Sent: Wednesday, January 02, 2008 10:21 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Incoming Calls



I am having a problem that I would like to verify if someone could help...I am using bandwith.com as my SIP TRUNK provider. When I place the phone number in the DID number field ( using Elastix) it gives me an error message stating the phone number I dialed is not in service. When I leave the DID number and CLID number blanks it works fine. I really need to have the system identifying multiple phone numbers ( multiple trunks ) but I have not been able to do so. Would anyone be able to help?



Thanks,



Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767



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jonnt at taylortelepho...
Guest





PostPosted: Wed Jan 02, 2008 12:25 pm    Post subject: [asterisk-users] Incoming Calls Reply with quote

allow=ulaw&alaw

canreinvite=no

context=from-internal

disallow=all

dtmfmode=auto

host=xxx.xxx.xxx.xxx (IP address)

insecure=very

nat=no

qualify=no

tos=none

type=peer



This should work for you. They only accept g711 and g729. There service only works with static ip's, so there is no auth used.



Jonn

_____

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paulo Pinheiro
Sent: Wednesday, January 02, 2008 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls



Hi Jose, I apologize for the lack of information..I am new to this...Let me try to be more specific:



I've got Asterisk installed on Linux. I am using Elastix as the front end to make changes in the system.



Under the Trunk set up these are my setting for the Peer Details:



allow=ulaw&alaw&gsm

auth=plaintext

canreinvite=no

context=from-internal

disallow=all

dtmfmode=inband

fromdomain=xxx.xxx.xxx.xxx (IP address)

host=xxx.xxx.xxx.xxx (IP address)

insecure=very

nat=no

qualify=no

tos=none

type=friend



these are my settings for User Details:



allow=ulaw

canreinvite=no

context=from-sip-external

dtmfmode=rfc2833

host=xxx.xxx.xxx.xxx (IP addres)

nat=no

port=5060

reinvite=no

type=peer



When setting up the income routes if I place the phone number in the DID Number field, when calling the number I receive a message stating the phone number is not listed or out of service. When I leave the DID Number field blank everything works because it does a catch all scenario but that is not what I am looking for.



I have tried to place the phone number with +1 in front of it and still does not work. Any way to help?



Thanks much,

Paulo





From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jose P. Espinal
Sent: Wednesday, January 02, 2008 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls



Hi Mr. Paulo,

Could you please explain this situation in a more detailed way to see how can we help you?

Regards,

Paulo Pinheiro wrote:

I am having a problem that I would like to verify if someone could help...I am using bandwith.com as my SIP TRUNK provider. When I place the phone number in the DID number field ( using Elastix) it gives me an error message stating the phone number I dialed is not in service. When I leave the DID number and CLID number blanks it works fine. I really need to have the system identifying multiple phone numbers ( multiple trunks ) but I have not been able to do so. Would anyone be able to help?



Thanks,



Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767











_____






_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users





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Paulo.Pinheiro at cent...
Guest





PostPosted: Wed Jan 02, 2008 1:20 pm    Post subject: [asterisk-users] Incoming Calls Reply with quote

Hi John, I have copied your changes in the Peer Details section of the
trunk set up...then I went ahead and added the DID number in the Income
Routes but still did not work. I tried the number alone and also tried
adding the + sign in front of it. Do you think we should have any
changes in the User Details section of the trunk set up?



Thanks much,

Paulo



From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonn R
Taylor
Sent: Wednesday, January 02, 2008 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls



allow=ulaw&alaw

canreinvite=no

context=from-internal

disallow=all

dtmfmode=auto

host=xxx.xxx.xxx.xxx (IP address)

insecure=very

nat=no

qualify=no

tos=none

type=peer



This should work for you. They only accept g711 and g729. There service
only works with static ip's, so there is no auth used.



Jonn

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paulo
Pinheiro
Sent: Wednesday, January 02, 2008 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls



Hi Jose, I apologize for the lack of information..I am new to this...Let
me try to be more specific:



I've got Asterisk installed on Linux. I am using Elastix as the front
end to make changes in the system.



Under the Trunk set up these are my setting for the Peer Details:



allow=ulaw&alaw&gsm

auth=plaintext

canreinvite=no

context=from-internal

disallow=all

dtmfmode=inband

fromdomain=xxx.xxx.xxx.xxx (IP address)

host=xxx.xxx.xxx.xxx (IP address)

insecure=very

nat=no

qualify=no

tos=none

type=friend



these are my settings for User Details:



allow=ulaw

canreinvite=no

context=from-sip-external

dtmfmode=rfc2833

host=xxx.xxx.xxx.xxx (IP addres)

nat=no

port=5060

reinvite=no

type=peer



When setting up the income routes if I place the phone number in the DID
Number field, when calling the number I receive a message stating the
phone number is not listed or out of service. When I leave the DID
Number field blank everything works because it does a catch all scenario
but that is not what I am looking for.



I have tried to place the phone number with +1 in front of it and still
does not work. Any way to help?



Thanks much,

Paulo





From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jose P.
Espinal
Sent: Wednesday, January 02, 2008 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Incoming Calls



Hi Mr. Paulo,

Could you please explain this situation in a more detailed way to see
how can we help you?

Regards,

Paulo Pinheiro wrote:

I am having a problem that I would like to verify if someone could
help...I am using bandwith.com as my SIP TRUNK provider. When I place
the phone number in the DID number field ( using Elastix) it gives me an
error message stating the phone number I dialed is not in service. When
I leave the DID number and CLID number blanks it works fine. I really
need to have the system identifying multiple phone numbers ( multiple
trunks ) but I have not been able to do so. Would anyone be able to
help?



Thanks,



Paulo Pinheiro

President

Centurion Vision Inc.

www.centurionvision.com

Phone: 800.714.8776 ext.103

Fax: 561.338.0767










________________________________








_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



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Tony at plack.net
Guest





PostPosted: Wed Jan 02, 2008 1:48 pm    Post subject: [asterisk-users] Incoming Calls Reply with quote

Quote:
Hi John, I have copied your changes in the Peer Details section of
the trunk set up?then I went ahead and added the DID number in the
Income Routes but still did not work. I tried the number alone and
also tried adding the + sign in front of it. Do you think we should
have any changes in the User Details section of the trunk set up?


Thanks much,

Paulo

Paulo,
I think I understand what you are trying to do, but that is not the correct location for it.

The DID number is the number calling you. If you place your trunk number into the configuration, you will only get calls on that trunk when the DID is your trunk number (not really correct).

What you need, is a dial-plan entry with _+15551234567,n,Noop type of setting in the extensions.conf. What you are changing is the sip.conf file.

I do not know if Elastix supports that type of change directly in the extensions.conf, but it should.

Tony Plack
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