[asterisk-users] Asterisk to be used with Ciscs media gateways

[Mohit Saxena]

Sip.comf

[PCCW-KPN]
type=peer
host=41.205.190.15
allow=ulaw
qualify=100
nat=no
canreinvite=no
user=07028000709


extension.conf
exten=07028XXXXXX,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:mohits at starcomms.com
www.starcomms.com


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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" <MohitS at starcomms.com> 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:mohits at starcomms.com
> www.starcomms.com
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at 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 <MohitS at starcomms.com>
> wrote:
> > I have been trying to do this since 2 days but couldn't make
> it....need your help..
>
> Well, you could certainly ask Cisco for help.
> You did pay Cisco money, right?
>
> > PSTN-----Cisco AS5350-------Asterisk Box--------VoIP 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.
>
> --
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