[asterisk-users] Asterisk to be used with Ciscs media gateways
Tim Nelson
tnelson at rockbochs.com
Tue Mar 16 09:02:23 CDT 2010
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" <MohitS at starcomms.com> wrote:
> Still no luck....
>
> Br,
> Mohit
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at 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=07028XXXXXX,1,Dial(SIP/${EXTEN}@PCCW-KPN)
>
> You aren't sending an outbound DID with just SIP/PCCW-KPN.
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at 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=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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