Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Sip.conf
[mohit]
type=friend
allow=ulaw
secret=mohit
callerid="Mohit" <1234>
host=dynamic
canreinvite=no
context=internal
nat=no
regexten=1234
[41.205.190.15]
type=peer
host=41.205.190.15
insecure=very
disallow=all
allow=ulaw
Extensions.conf
exten = 1234,1,Dial(SIP/mohit)
;exten = 07028063180,1,Dial(SIP/PCCW-KPN)
;exten => _07028XX,1,Dial(SIP/PCCW-KPN/${EXTEN})
;exten => _07028XX,1,Macro(page,SIP/${EXTEN})
;exten => _0702X.,1,Dial(SIP/${ext...@pccw-kpn)
exten => _07028XX,1,Dial(SIP/${ext...@41.205.190.15)
I am able to place a call from cisco side towards extension 1234. But while I
am trying to place a call from extension 1234 to any PSTN number with paatern
07028XX, the call drops and the sip log on asterisk box says
[Mar 16 16:19:20] NOTICE[5504] chan_sip.c: Call from 'mohit' to extension
'07028000709' rejected because extension not found.
Br,
Mohit
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Tuesday, March 16, 2010 2:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Under the dial-peer on the Cisco change these two things:
no incoming called-number .T
destination-pattern 07028X (use the actual number to match or .'s for
wildcard)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohit Saxena
Sent: Tuesday, March 16, 2010 2:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Still no luck
Br,
Mohit
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Monday, March 15, 2010 6:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
exten=07028XX,1,Dial(SIP/${ext...@pccw-kpn)
You aren't sending an outbound DID with just SIP/PCCW-KPN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohit Saxena
Sent: Monday, March 15, 2010 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Sip.comf
[PCCW-KPN]
type=peer
host=41.205.190.15
allow=ulaw
qualify=100
nat=no
canreinvite=no
user=07028000709
extension.conf
exten=07028XX,1,Dial(SIP/PCCW-KPN)
Cisco Gateway:
dial-peer voice 110 voip
description Voip peer to test the server
destination-pattern 1234
session protocol sipv2
session target ipv4:196.3.60.24
session transport udp
incoming called-number .T
dtmf-relay rtp-nte
codec g711ulaw
fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 1
hs-redundancy 1 fallback pass-through g711ulaw
clid strip
Br,
Mohit C. Saxena I Data/ISP Department
Starcomms Plc.
1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709
email:moh...@starcomms.com
www.starcomms.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, March 15, 2010 6:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Continuing with the top posting parade...
Can you post your {sanitized} sip.conf and your extensions.conf for
inspection?
--Tim
- "Mohit Saxena" wrote:
> The problem is not with cisco as the SIP header on debug doesn't
> contain the called number. It only says To:sip:ip add of cisco gw. It
> should say number:ip addr of cisco gw.
>
> Br,
> Mohit C. Saxena I Data/ISP Department
> Starcomms Plc.
> 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,
> +234-702-8000-709 email:moh...@starcomms.com
> www.starcomms.com
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Backeberg
> Sent: Monday, March 15, 2010 5:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena
> wrote:
> > I have been trying to do this since 2 days but couldn't make
> itneed your help..
>
> Well, you could certainly ask Cisco for help.
> You did pay Cisco money, right?
>
> > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
>
> > I am able to place call from cisco gateway to the asterisk box and
>
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
More top posting goodness...
Please post your updated dialplan. After making the change, did you
reload/restart Asterisk so the changes would take effect?
--Tim
- "Mohit Saxena" wrote:
> Still no luck
>
> Br,
> Mohit
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
> Sent: Monday, March 15, 2010 6:54 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> exten=07028XX,1,Dial(SIP/${ext...@pccw-kpn)
>
> You aren't sending an outbound DID with just SIP/PCCW-KPN.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohit
> Saxena
> Sent: Monday, March 15, 2010 12:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> Sip.comf
>
> [PCCW-KPN]
> type=peer
> host=41.205.190.15
> allow=ulaw
> qualify=100
> nat=no
> canreinvite=no
> user=07028000709
>
>
> extension.conf
> exten=07028XX,1,Dial(SIP/PCCW-KPN)
>
> Cisco Gateway:
> dial-peer voice 110 voip
> description Voip peer to test the server
> destination-pattern 1234
> session protocol sipv2
> session target ipv4:196.3.60.24
> session transport udp
> incoming called-number .T
> dtmf-relay rtp-nte
> codec g711ulaw
> fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy
> 1
> hs-redundancy 1 fallback pass-through g711ulaw
> clid strip
>
> Br,
> Mohit C. Saxena I Data/ISP Department
> Starcomms Plc.
> 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,
> +234-702-8000-709
> email:moh...@starcomms.com
> www.starcomms.com
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim
> Nelson
> Sent: Monday, March 15, 2010 6:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> Continuing with the top posting parade...
>
> Can you post your {sanitized} sip.conf and your extensions.conf for
> inspection?
>
> --Tim
>
> - "Mohit Saxena" wrote:
> > The problem is not with cisco as the SIP header on debug doesn't
> > contain the called number. It only says To:sip:ip add of cisco gw.
> It
> > should say number:ip addr of cisco gw.
> >
> > Br,
> > Mohit C. Saxena I Data/ISP Department
> > Starcomms Plc.
> > 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,
> > +234-702-8000-709 email:moh...@starcomms.com
> > www.starcomms.com
> >
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> > Backeberg
> > Sent: Monday, March 15, 2010 5:48 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> > gateways
> >
> > On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena
>
> > wrote:
> > > I have been trying to do this since 2 days but couldn't make
> > itneed your help..
> >
> > Well, you could certainly ask Cisco for help.
> > You did pay Cisco money, right?
> >
> > > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
> >
> > > I am able to place call from cisco gateway to the asterisk box
> and
> > also to some softphones extensions but >when making a call from
> > softphone from asterisk box to PSTN, it fails. While I debug on
> Cisco
> > gateway I found >that the To field is SIP header is coming as
> > sip:41.205.190.15 which is not correct, instead it should be dialed
> > >number:41.205.190.15
> >
> > Then the problem seems to be between your asterisk box and your
> > Cisco.
> > Perhaps if you told us what you were trying to SIP dial, we would
> be
> > able to tell us what you did wrong.
> >
> > > Has any one of you tried using Asterisk in this scenario
> >
> > yes.
> >
> > > and also to do LCR and Quality based routing of International
> > calls?
> >
> > I don't know what that means.
> >
> > > Please let me know if there is any documentation /example of this
> > kind available
> >
> > There is.
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Under the dial-peer on the Cisco change these two things:
no incoming called-number .T
destination-pattern 07028X (use the actual number to match or .'s for
wildcard)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohit Saxena
Sent: Tuesday, March 16, 2010 2:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Still no luck
Br,
Mohit
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Monday, March 15, 2010 6:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
exten=07028XX,1,Dial(SIP/${ext...@pccw-kpn)
You aren't sending an outbound DID with just SIP/PCCW-KPN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohit Saxena
Sent: Monday, March 15, 2010 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Sip.comf
[PCCW-KPN]
type=peer
host=41.205.190.15
allow=ulaw
qualify=100
nat=no
canreinvite=no
user=07028000709
extension.conf
exten=07028XX,1,Dial(SIP/PCCW-KPN)
Cisco Gateway:
dial-peer voice 110 voip
description Voip peer to test the server
destination-pattern 1234
session protocol sipv2
session target ipv4:196.3.60.24
session transport udp
incoming called-number .T
dtmf-relay rtp-nte
codec g711ulaw
fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 1
hs-redundancy 1 fallback pass-through g711ulaw
clid strip
Br,
Mohit C. Saxena I Data/ISP Department
Starcomms Plc.
1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709
email:moh...@starcomms.com
www.starcomms.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, March 15, 2010 6:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Continuing with the top posting parade...
Can you post your {sanitized} sip.conf and your extensions.conf for
inspection?
--Tim
- "Mohit Saxena" wrote:
> The problem is not with cisco as the SIP header on debug doesn't
> contain the called number. It only says To:sip:ip add of cisco gw. It
> should say number:ip addr of cisco gw.
>
> Br,
> Mohit C. Saxena I Data/ISP Department
> Starcomms Plc.
> 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,
> +234-702-8000-709 email:moh...@starcomms.com
> www.starcomms.com
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Backeberg
> Sent: Monday, March 15, 2010 5:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena
> wrote:
> > I have been trying to do this since 2 days but couldn't make
> itneed your help..
>
> Well, you could certainly ask Cisco for help.
> You did pay Cisco money, right?
>
> > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
>
> > I am able to place call from cisco gateway to the asterisk box and
> also to some softphones extensions but >when making a call from
> softphone from asterisk box to PSTN, it fails. While I debug on Cisco
> gateway I found >that the To field is SIP header is coming as
> sip:41.205.190.15 which is not correct, instead it should be dialed
> >number:41.205.190.15
>
> Then the problem seems to be between your asterisk box and your
> Cisco.
> Perhaps if you told us what you were trying to SIP dial, we would be
> able to tell us what you did wrong.
>
> > Has any one of you tried using Asterisk in this scenario
>
> yes.
>
> > and also to do LCR and Quality based routing of International
> calls?
>
> I don't know what that means.
>
> > Please let me know if there is any documentation /example of this
> kind available
>
> There is.
> cisco.com
> you pay them, then you can use their documentation.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCR
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Still no luck
Br,
Mohit
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Monday, March 15, 2010 6:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
exten=07028XX,1,Dial(SIP/${ext...@pccw-kpn)
You aren't sending an outbound DID with just SIP/PCCW-KPN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohit Saxena
Sent: Monday, March 15, 2010 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Sip.comf
[PCCW-KPN]
type=peer
host=41.205.190.15
allow=ulaw
qualify=100
nat=no
canreinvite=no
user=07028000709
extension.conf
exten=07028XX,1,Dial(SIP/PCCW-KPN)
Cisco Gateway:
dial-peer voice 110 voip
description Voip peer to test the server
destination-pattern 1234
session protocol sipv2
session target ipv4:196.3.60.24
session transport udp
incoming called-number .T
dtmf-relay rtp-nte
codec g711ulaw
fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 1
hs-redundancy 1 fallback pass-through g711ulaw
clid strip
Br,
Mohit C. Saxena I Data/ISP Department
Starcomms Plc.
1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709
email:moh...@starcomms.com
www.starcomms.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, March 15, 2010 6:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Continuing with the top posting parade...
Can you post your {sanitized} sip.conf and your extensions.conf for
inspection?
--Tim
- "Mohit Saxena" wrote:
> The problem is not with cisco as the SIP header on debug doesn't
> contain the called number. It only says To:sip:ip add of cisco gw. It
> should say number:ip addr of cisco gw.
>
> Br,
> Mohit C. Saxena I Data/ISP Department
> Starcomms Plc.
> 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,
> +234-702-8000-709 email:moh...@starcomms.com
> www.starcomms.com
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Backeberg
> Sent: Monday, March 15, 2010 5:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena
> wrote:
> > I have been trying to do this since 2 days but couldn't make
> itneed your help..
>
> Well, you could certainly ask Cisco for help.
> You did pay Cisco money, right?
>
> > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
>
> > I am able to place call from cisco gateway to the asterisk box and
> also to some softphones extensions but >when making a call from
> softphone from asterisk box to PSTN, it fails. While I debug on Cisco
> gateway I found >that the To field is SIP header is coming as
> sip:41.205.190.15 which is not correct, instead it should be dialed
> >number:41.205.190.15
>
> Then the problem seems to be between your asterisk box and your
> Cisco.
> Perhaps if you told us what you were trying to SIP dial, we would be
> able to tell us what you did wrong.
>
> > Has any one of you tried using Asterisk in this scenario
>
> yes.
>
> > and also to do LCR and Quality based routing of International
> calls?
>
> I don't know what that means.
>
> > Please let me know if there is any documentation /example of this
> kind available
>
> There is.
> cisco.com
> you pay them, then you can use their documentation.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailm
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
- "Mohit Saxena" wrote:
> extension.conf
> exten=07028XX,1,Dial(SIP/PCCW-KPN)
Here is your issue. Shouldn't you be sending the number you'd like to dial with
the call? Try this:
exten => 07028XX,1,Dial(SIP/PCCW-KPN/${EXTEN})
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
exten=07028XX,1,Dial(SIP/${ext...@pccw-kpn)
You aren't sending an outbound DID with just SIP/PCCW-KPN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohit Saxena
Sent: Monday, March 15, 2010 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Sip.comf
[PCCW-KPN]
type=peer
host=41.205.190.15
allow=ulaw
qualify=100
nat=no
canreinvite=no
user=07028000709
extension.conf
exten=07028XX,1,Dial(SIP/PCCW-KPN)
Cisco Gateway:
dial-peer voice 110 voip
description Voip peer to test the server
destination-pattern 1234
session protocol sipv2
session target ipv4:196.3.60.24
session transport udp
incoming called-number .T
dtmf-relay rtp-nte
codec g711ulaw
fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 1
hs-redundancy 1 fallback pass-through g711ulaw
clid strip
Br,
Mohit C. Saxena I Data/ISP Department
Starcomms Plc.
1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709
email:moh...@starcomms.com
www.starcomms.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, March 15, 2010 6:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Continuing with the top posting parade...
Can you post your {sanitized} sip.conf and your extensions.conf for
inspection?
--Tim
- "Mohit Saxena" wrote:
> The problem is not with cisco as the SIP header on debug doesn't
> contain the called number. It only says To:sip:ip add of cisco gw. It
> should say number:ip addr of cisco gw.
>
> Br,
> Mohit C. Saxena I Data/ISP Department
> Starcomms Plc.
> 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,
> +234-702-8000-709 email:moh...@starcomms.com
> www.starcomms.com
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Backeberg
> Sent: Monday, March 15, 2010 5:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena
> wrote:
> > I have been trying to do this since 2 days but couldn't make
> itneed your help..
>
> Well, you could certainly ask Cisco for help.
> You did pay Cisco money, right?
>
> > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
>
> > I am able to place call from cisco gateway to the asterisk box and
> also to some softphones extensions but >when making a call from
> softphone from asterisk box to PSTN, it fails. While I debug on Cisco
> gateway I found >that the To field is SIP header is coming as
> sip:41.205.190.15 which is not correct, instead it should be dialed
> >number:41.205.190.15
>
> Then the problem seems to be between your asterisk box and your
> Cisco.
> Perhaps if you told us what you were trying to SIP dial, we would be
> able to tell us what you did wrong.
>
> > Has any one of you tried using Asterisk in this scenario
>
> yes.
>
> > and also to do LCR and Quality based routing of International
> calls?
>
> I don't know what that means.
>
> > Please let me know if there is any documentation /example of this
> kind available
>
> There is.
> cisco.com
> you pay them, then you can use their documentation.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Yes, I mean the same Least Cost routing. Br, Mohit C. Saxena I Data/ISP Department Starcomms Plc. 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709 email:moh...@starcomms.com www.starcomms.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Monday, March 15, 2010 6:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways > and also to do LCR and Quality based routing of International calls? I don't know what that means. LCR = "Least Cost Routing" Routing a call based on the quality or cost of a route (PSTN term vs SIP to PSTN term vs SIP to SIP) is actually quite common. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The information contained in this message (including any attachments) is confidential and may be privileged. If you have received it by mistake please notify the sender by return e-mail and permanently delete this message and any attachments from your system. Any form of dissemination, use, review, distribution, printing or copying of this message in whole or in part is strictly prohibited if you are not the intended recipient of this e-mail. Please note that e-mails are susceptible to change. STARCOMMS PLC shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt or damage to your system. STARCOMMS PLC does not guarantee that the integrity of this communication has been maintained or that this communication is free of viruses, interceptions or interferences. STARCOMMS PLC reserves the right to monitor all e-mail communications, whether related to the business of STARCOMMS or not, through its internal or external networks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Sip.comf
[PCCW-KPN]
type=peer
host=41.205.190.15
allow=ulaw
qualify=100
nat=no
canreinvite=no
user=07028000709
extension.conf
exten=07028XX,1,Dial(SIP/PCCW-KPN)
Cisco Gateway:
dial-peer voice 110 voip
description Voip peer to test the server
destination-pattern 1234
session protocol sipv2
session target ipv4:196.3.60.24
session transport udp
incoming called-number .T
dtmf-relay rtp-nte
codec g711ulaw
fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 1
hs-redundancy 1 fallback pass-through g711ulaw
clid strip
Br,
Mohit C. Saxena I Data/ISP Department
Starcomms Plc.
1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709
email:moh...@starcomms.com
www.starcomms.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, March 15, 2010 6:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Continuing with the top posting parade...
Can you post your {sanitized} sip.conf and your extensions.conf for inspection?
--Tim
- "Mohit Saxena" wrote:
> The problem is not with cisco as the SIP header on debug doesn't
> contain the called number. It only says To:sip:ip add of cisco gw. It
> should say number:ip addr of cisco gw.
>
> Br,
> Mohit C. Saxena I Data/ISP Department
> Starcomms Plc.
> 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,
> +234-702-8000-709 email:moh...@starcomms.com
> www.starcomms.com
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Backeberg
> Sent: Monday, March 15, 2010 5:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena
> wrote:
> > I have been trying to do this since 2 days but couldn't make
> itneed your help..
>
> Well, you could certainly ask Cisco for help.
> You did pay Cisco money, right?
>
> > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
>
> > I am able to place call from cisco gateway to the asterisk box and
> also to some softphones extensions but >when making a call from
> softphone from asterisk box to PSTN, it fails. While I debug on Cisco
> gateway I found >that the To field is SIP header is coming as
> sip:41.205.190.15 which is not correct, instead it should be dialed
> >number:41.205.190.15
>
> Then the problem seems to be between your asterisk box and your
> Cisco.
> Perhaps if you told us what you were trying to SIP dial, we would be
> able to tell us what you did wrong.
>
> > Has any one of you tried using Asterisk in this scenario
>
> yes.
>
> > and also to do LCR and Quality based routing of International
> calls?
>
> I don't know what that means.
>
> > Please let me know if there is any documentation /example of this
> kind available
>
> There is.
> cisco.com
> you pay them, then you can use their documentation.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
DISCLAIMER: The information contained in this message (including any
attachments) is confidential and may be privileged. If you have received it by
mistake please notify the sender by return e-mail and permanently delete this
message and any attachments from your system. Any form of dissemination, use,
review, distribution, printing or copying of this message in whole or i
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
> and also to do LCR and Quality based routing of International calls? I don't know what that means. LCR = "Least Cost Routing" Routing a call based on the quality or cost of a route (PSTN term vs SIP to PSTN term vs SIP to SIP) is actually quite common. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
On Mon, 15 Mar 2010, Mohit Saxena wrote: > We are a mobile operator so has to work with the PSTN side E1s from the > Mobile switch. This is the reason for using Cisco Media gateways. I know you may be stuck with them, but you could just as easily plug in a Digium/Sangoma/Rhino T1/E1 card (or Xorcom channel bank?) into your asterisk box and you would be able to accomplish the same thing, but in IMO a much more asterisk-friendly way. Can't help you with the Cisco config... you will need to post a lot more details about your asterisk config if you want help on that side. j > > Kindly help > > Br, > Mohit C. Saxena I Data/ISP Department > Starcomms Plc. > 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709 > email:moh...@starcomms.com > www.starcomms.com > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere > Sent: Monday, March 15, 2010 6:06 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways > > > > On Mon, 15 Mar 2010, David Backeberg wrote: > >> >>> and also to do LCR and Quality based routing of International calls? >> >> I don't know what that means. >> > > Least Cost Routing. Asterisk doesn't have anything built in for this. We > do it with an in-house AGI. Others have done similar things that you > might be able to buy. Try on asterisk-biz. > > The question I have is - why the Cisco? Assuming you have SIP or H.323 > capable phones, just dump the Cisco and use the asterisk box for the whole > shebang. > > j > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
Continuing with the top posting parade...
Can you post your {sanitized} sip.conf and your extensions.conf for inspection?
--Tim
- "Mohit Saxena" wrote:
> The problem is not with cisco as the SIP header on debug doesn't
> contain the called number. It only says To:sip:ip add of cisco gw. It
> should say number:ip addr of cisco gw.
>
> Br,
> Mohit C. Saxena I Data/ISP Department
> Starcomms Plc.
> 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,
> +234-702-8000-709 email:moh...@starcomms.com
> www.starcomms.com
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
> Backeberg
> Sent: Monday, March 15, 2010 5:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media
> gateways
>
> On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena
> wrote:
> > I have been trying to do this since 2 days but couldn't make
> itneed your help..
>
> Well, you could certainly ask Cisco for help.
> You did pay Cisco money, right?
>
> > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
>
> > I am able to place call from cisco gateway to the asterisk box and
> also to some softphones extensions but >when making a call from
> softphone from asterisk box to PSTN, it fails. While I debug on Cisco
> gateway I found >that the To field is SIP header is coming as
> sip:41.205.190.15 which is not correct, instead it should be dialed
> >number:41.205.190.15
>
> Then the problem seems to be between your asterisk box and your
> Cisco.
> Perhaps if you told us what you were trying to SIP dial, we would be
> able to tell us what you did wrong.
>
> > Has any one of you tried using Asterisk in this scenario
>
> yes.
>
> > and also to do LCR and Quality based routing of International
> calls?
>
> I don't know what that means.
>
> > Please let me know if there is any documentation /example of this
> kind available
>
> There is.
> cisco.com
> you pay them, then you can use their documentation.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
We are a mobile operator so has to work with the PSTN side E1s from the Mobile switch. This is the reason for using Cisco Media gateways. Kindly help Br, Mohit C. Saxena I Data/ISP Department Starcomms Plc. 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709 email:moh...@starcomms.com www.starcomms.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Monday, March 15, 2010 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways On Mon, 15 Mar 2010, David Backeberg wrote: > >> and also to do LCR and Quality based routing of International calls? > > I don't know what that means. > Least Cost Routing. Asterisk doesn't have anything built in for this. We do it with an in-house AGI. Others have done similar things that you might be able to buy. Try on asterisk-biz. The question I have is - why the Cisco? Assuming you have SIP or H.323 capable phones, just dump the Cisco and use the asterisk box for the whole shebang. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
The problem is not with cisco as the SIP header on debug doesn't contain the called number. It only says To:sip:ip add of cisco gw. It should say number:ip addr of cisco gw. Br, Mohit C. Saxena I Data/ISP Department Starcomms Plc. 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709 email:moh...@starcomms.com www.starcomms.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Monday, March 15, 2010 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena wrote: > I have been trying to do this since 2 days but couldn't make itneed your > help.. Well, you could certainly ask Cisco for help. You did pay Cisco money, right? > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers > I am able to place call from cisco gateway to the asterisk box and also to > some softphones extensions but >when making a call from softphone from > asterisk box to PSTN, it fails. While I debug on Cisco gateway I found >that > the To field is SIP header is coming as sip:41.205.190.15 which is not > correct, instead it should be dialed >number:41.205.190.15 Then the problem seems to be between your asterisk box and your Cisco. Perhaps if you told us what you were trying to SIP dial, we would be able to tell us what you did wrong. > Has any one of you tried using Asterisk in this scenario yes. > and also to do LCR and Quality based routing of International calls? I don't know what that means. > Please let me know if there is any documentation /example of this kind > available There is. cisco.com you pay them, then you can use their documentation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
On Mon, 15 Mar 2010, David Backeberg wrote: > >> and also to do LCR and Quality based routing of International calls? > > I don't know what that means. > Least Cost Routing. Asterisk doesn't have anything built in for this. We do it with an in-house AGI. Others have done similar things that you might be able to buy. Try on asterisk-biz. The question I have is - why the Cisco? Assuming you have SIP or H.323 capable phones, just dump the Cisco and use the asterisk box for the whole shebang. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena wrote: > I have been trying to do this since 2 days but couldn't make itneed your > help.. Well, you could certainly ask Cisco for help. You did pay Cisco money, right? > PSTN-Cisco AS5350---Asterisk BoxVoIP Providers > I am able to place call from cisco gateway to the asterisk box and also to > some softphones extensions but >when making a call from softphone from > asterisk box to PSTN, it fails. While I debug on Cisco gateway I found >that > the To field is SIP header is coming as sip:41.205.190.15 which is not > correct, instead it should be dialed >number:41.205.190.15 Then the problem seems to be between your asterisk box and your Cisco. Perhaps if you told us what you were trying to SIP dial, we would be able to tell us what you did wrong. > Has any one of you tried using Asterisk in this scenario yes. > and also to do LCR and Quality based routing of International calls? I don't know what that means. > Please let me know if there is any documentation /example of this kind > available There is. cisco.com you pay them, then you can use their documentation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk to be used with Ciscs media gateways
Hello Guys, I have been trying to do this since 2 days but couldn't make itneed your help.. The scenario is as under: PSTN-Cisco AS5350---Asterisk BoxVoIP Providers I am trying to use SIP on Cisco Gateways and Asterisk box for the connection. The configuration is as under: Sip.comf [PCCW-KPN] type=peer host=41.205.190.15 allow=ulaw qualify=100 nat=no canreinvite=no user=07028000709 extension.conf exten=07028XX,1,Dial(SIP/PCCW-KPN) Cisco Gateway: dial-peer voice 110 voip description Voip peer to test the server destination-pattern 1234 session protocol sipv2 session target ipv4:196.3.60.24 session transport udp incoming called-number .T dtmf-relay rtp-nte codec g711ulaw fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 1 hs-redundancy 1 fallback pass-through g711ulaw clid strip I am able to place call from cisco gateway to the asterisk box and also to some softphones extensions but when making a call from softphone from asterisk box to PSTN, it fails. While I debug on Cisco gateway I found that the To field is SIP header is coming as sip:41.205.190.15 which is not correct, instead it should be dialed number:41.205.190.15 Has any one of you tried using Asterisk in this scenario and also to do LCR and Quality based routing of International calls? Please let me know if there is any documentation /example of this kind available/ Br, Mohit DISCLAIMER: The information contained in this message (including any attachments) is confidential and may be privileged. If you have received it by mistake please notify the sender by return e-mail and permanently delete this message and any attachments from your system. Any form of dissemination, use, review, distribution, printing or copying of this message in whole or in part is strictly prohibited if you are not the intended recipient of this e-mail. Please note that e-mails are susceptible to change. STARCOMMS PLC shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt or damage to your system. STARCOMMS PLC does not guarantee that the integrity of this communication has been maintained or that this communication is free of viruses, interceptions or interferences. STARCOMMS PLC reserves the right to monitor all e-mail communications, whether related to the business of STARCOMMS or not, through its internal or external networks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
