[Previous] Issues with Connection Rules: matching by inbound number - ...

 [Next] Apple Bloog » Blog Archive » ZoIPer 2.12 - SIP and...

Some related posts from Technorati and Google.

[Dialogic Exchange Network] Why is dial tone being detected and call being hung up after 1 ...: /so/so_ptli.c:696 SoLiHitConReq(pst, spId(0), suConId(19), remAddr.type(4), remAddr.port(5060), remAddr.address(C0A80150),localAddr.type(4), localAddr.port(0), localAddr.address(C0A8013C), tpar, srvcType (0)) .

[Voxalot / SIP Broker Support Forums] Problem calling another Voxalot - Voxalot / SIP Broker Support Forums: is a feature that can be activated only when there is a fault that causes a non-routable LAN IP address to be used as the Contact Address in a SIP SDP packet. This can happen when the ATA is behind a symmetric NAT router, the ATA is mis-configured, or a STUN server is not working, etc.

[Internet Drafts: sip] draft-ietf-sip-199-08 - Response Code for Indication of Terminated ...: Since all SIP entities involved in a session setup do not necessarily support the specific meaning of the 199 Early Dialog Terminated provisional response, the sender of the response MUST be prepared to receive SIP requests and responses associated with the dialog for which the 199 response was sent (a proxy can receive SIP messages from either direction). If such request is received by a UA, it MUST act in the same way as if it had received the request after sending the final non-2xx response to the INVITE, as specified in [RFC3261].

[TechNet Blogs] Microsoft UK UC Blog : Reverse Number Lookup and Dealing with ...: When the invite message is received by the Office Communicator client it uses either the “From” field or the “P-Asserted-Identity” field to match against the address book and local Outlook contacts. The problem here is that the numbers in the address book come from Active Directory and these are usually in the short dial format as they are used by users who are not using OCS enterprise voice.

[New Current Internet Drafts (All Categories)] draft-levy-sip-diversion-10 - Diversion Indication in SIP: Internet-Draft Diversion header July 2009 SIP proxy server 3 (NightService.com) to UAS1 (Carol): INVITE sip:carol@uas1.nightservice.com SIP/2.0 Via: SIP/2.0/UDP p3.isp.com Via: SIP/2.0/UDP p2.isp.com Via: SIP/2.0/UDP p1.isp.com Via: ....

[maemo.org - Talk] How-to: Google Voice Workaround [In-depth] - maemo.org - Talk: Plus you may need input a GV number as a forward number and it will infact tell you that it is a GV #. (In the future sneaky people such as myself will be able to determine if a number is a GV # simply by entering it as registration forward # as well as if a physical line is attached to a GV # by entering it a forward #).

[New Current Internet Drafts (All Categories)] draft-zourzouvillys-sip-via-cookie-02 - to be used as an amplifier ...: Even if it were a SIP stack and it knew it was being spoofed (for example, due to the received parameter in the only via of a response matching its public IP address), this does not solve non-INVITE transaction problems. 4XX Via Cookie Required The IANA is requested to register a new SIP response code which is described in section X.

[Dialogic Exchange Network] Brand new to SIP, DMG 1000 and entire VoIP - very overwhelmed and ...: What we need to do is eventually configure a normal PSTN telephone number somewhere in our PC software (that we will have to write for ourselves, a client side application) and then have our PC software call this number, which presumably will talk SIP to the DMG, and then the DMG will dial out and communicate through the PSTN. I sure hope that I have that flow figured out at least, but like I said, I'm in acroymn overload.

[Mu Dynamics Research Labs] Mu Dynamics Research Labs » Blog Archive » D/DoS Testing Network ...: If you look at the SIP INVITE message there are a number of regions within the payload that either need to be randomized or need to be tied to the L2/L3/L4 header values. In other words if we are generating random IP addresses and/or ports for each packet, then the payload needs to reflect that to make the server think it’s unique and valid.

[Voxeo Labs] CTO Lab » Blog Archive » Yammer, Present.ly, Laconica and pushing ...: I'll note that anyone can create a network and then invite anyone into that network, so in many respects it's not much different from Yammer, except that you don' t need a company email address and you have administrative powers from the beginning. To that point, the person creating the “network” starts off with full administrative powers and can also give others administration privileges.

[Jan Luehe's Blog] Jan Luehe's Blog : Weblog: This blog explains the motivation behind a converged application, provides a step-by-step analysis of how a ConvergedHttpSession may be used to initiate and terminate a SIP call from the HTTP servlet of a converged application, explains how the routing decisions of SailFin's Converged Loadbalancer influence the session id of a SipApplicationSession created from a ConvergedHttpSession, and finally shows how a ConvergedHttpSession fits into SailFin's in-memory replication framework that provides high-availability of HTTP and SIP artifacts in a SailFin cluster.

[The Dogg Blog] Best Business Tools: Google Voice | The Dogg Blog: Additionally, Gizmo5 customers can also forward calls to their Gizmo SIP number which can be answered using a free Gizmo5 client, Gizmocall (flash based client) or forwarded to any GoogleTalk, Skype or other SIP address so any VOIP client can be used.

[Pushkar bhatkoti's blog...........] CME SIP Trunking Configuration Example « Pushkar bhatkoti's blog”¦”¦”¦..: voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target sip-server dtmf-relay rtp-nte no vad!!!dial-peer voice 4 voip description **Outgoing Call to SIP Trunk** translation-profile outgoing PSTN_Outgoing destination-pattern 9[0-1][2-9]..[2-9]....voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target sip-server dtmf-relay rtp-nte no vad!!!dial-peer voice 5 voip description **911 Outgoing Call to SIP Trunk** translation-profile outgoing PSTN_Outgoing destination-pattern 911 voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target sip-server dtmf-relay rtp-nte no vad!!!dial-peer voice 6 voip description **Emergency Outgoing Call to SIP Trunk** translation-profile outgoing PSTN_Outgoing destination-pattern 9911 voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target sip-server dtmf-relay rtp-nte no vad!!!dial-peer voice 7 voip description **911/411 Outgoing Call to SIP Trunk** translation-profile outgoing PSTN_Outgoing destination-pattern 9[2-9]11 voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target sip-server dtmf-relay rtp-nte no vad!!!dial-peer voice 8 voip description **International Outgoing Call to SIP Trunk** translation-profile outgoing PSTN_Outgoing destination-pattern 9011T voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target sip-server dtmf-relay rtp-nte no vad!!!dial-peer voice 9 voip description **Star Code to SIP Trunk** destination-pattern *..

[Enterprise Tech Tips] Converged Enterprise Applications : Enterprise Tech Tips: One of the concepts introduced in the earlier tip is that of a converged application, that is, an application that includes both SIP and HTTP servlets. In this tip, you'll learn about converged enterprise applications and how the SIP Servlet API v1.1 specification simplifies the development of converged enterprise applications.

Reflected tags on Technorati: Blog, Sip Code, VoIPBlogging