we are in a Microsoft OCS SIP and VOIP environment. we are having a couple of issues with our unit. #1- out bound calls to the PSTN will auto disconnect after approx. 60 seconds. calls to OCS or VOIP phone numbers have no problems. #2- Inbound calls do not ring the analog phone; however,...
I have a switchvox AA350 SMB server in my organization which I run. Recently I aquired analog lines for external calls. I put two of the lines into the switchvox (the server came with a four port anolg card)I configured the lines and I was able to make calls through my SIP phones.Yesterday I came...
hi. I SIP implemented in Peer to Peer under I wanted to use windows(visual C++).Under its Linux site in the existing www.p2psip.org but i need something like that but under use windows.please you guide me to hurry. by.
I want to test SIP vulnerabilities using Sipp Message generating tool. i have XML files for generating messages. I have sipp running. But i am not able to execute below command. Run sipp with embedded server (uas) scenario: # ./sipp -sn uas Can i get any help. Thank you Regards Sindhu
I want to prove SIP vulnerabilities using SiVuS (The VoIp vulnerability scanner ) tool. The test Bed contains Asterisk server and two xlite clients. When Two xlite clients are registered with server. I am able to Remove their registrations from server by generating REGISTER SIP messages using...
How do I configure a port on an AudioCodes sip device to handle a analog modem?
dear All, I’m all new to VoIP and telephony in general and don't know the correct terminology...probably someone else already posted my questions in another thread but I managed to confuse the search engine with my query terms. Anyways... Basically I have a PSTN telephone line on which an ADSL...
We have one site having Cisco UC520 for IP telephony and another site we have nortel PBX. So is it possible to integrate both ? if yes then how and which features will work and which will not work ?? It would be great if someone help me out for the same.
Hi There, Anybody can help me to know the difference between Callmanager and SIP server? Call processing difference.. and protocols used. Thanks and regards, Shaju
Is a UAS required to use the same To: tag in a 487 response as was used in the 180/187. Scenario: A user INVITES B, B sends 183 (with To: tag filled in), then user A CANCELS B, B sends 487 with different To: tag
I'm registering my SIP UA (sipua1@192.168.1.2) with a proxy (asterisk). Asterisk has dial plans and when a call comes into one of the numbers it is forwarded to my UA as 1234@192.168.1.2. My application is an IVR that is supposed to handle multiple numbers like (1234, 4321 etc.). The IT guys...
I'm not able to configure SIP server in Cisco 3660 router (IOS is 12.4T). Can anyone please tell me how? Thank you.
I would like to setup a Innovaphone IP302 between a KIRK1500 DECT system and a Nortel CS1000 version 5.5.12 with SIP The communication between the Kirk1500 and the IP302 is OK The connection between the IP302 and the CS1000 is the problem! I receive the message forbidden on the CS1000, when I...
We are deploying a large scale VoIP system based on Asterisk and SIP, but some of our warehouses want to keep using the existing analog phones. Are there any large scale SIP based analog/IP converters out there? Or should we be looking for a different approach?
Hello, I am looking for expert advise on dynamic resource allocation in VoIP... I am trying to come up with some idea for using of dynamic badwidth adaptations for SIP teleconferencing in mobile wireless net? Would it be possible to use RTCP messagas to communicate resource availability ( bandwidth...
We are working to expand the Phone system to enterprise level over an existing MLPS network using SIP to integrate 5 different CTI applications. Would like mix of vendor neutral and vendor-specific content. This request for help was originally submitted to the Research Assistant on WhatIs.com.
Can SIP act as both a Proxy and B2BUA? If so, what type of calls require Porxy, and what type require B2BUA?
We are deploying a large scale VoIP system (+/- 5000 phones), based on Asterisk and SIP. In some of our warehouses we want to keep using the existing analog phones. We have 2 options (that we know of) to connect these analog phones to our VoIP system: 1) Linksys SPA2102 analog/IP convertor ...


