VoIP Mailing List Archives

[support at drdos.info]

Paulo Pinheiro wrote:

[rakh at dangerclan.net]

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:

[Tony at plack.net]

[Paulo.Pinheiro at cent...]

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






________________________________




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[jonnt at taylortelepho...]

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...]

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...]

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]