Tim Nelson
2010-Mar-16 14:02 UTC
[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" <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. > > > > -- > > > _____________________________________________________________________ > > -- 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 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 > > > -- > _____________________________________________________________________ > -- 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
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