Hello Guys, I have been trying to do this since 2 days but couldn't make it....need your help.. The scenario is as under:
PSTN-----Cisco AS5350-------Asterisk Box--------VoIP 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=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 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
