similar to: Sip communicator issue

Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but

Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but

Hi all, Thanks for your reply, I'm using sip communicator(in java that is intergrated with one ERP ) and asterisk is interfaced with this. i'm able to make calls between pingtel and Voip user, and also i can able to make call from Sip communicator to pingtel or Voip phone. but now i'm can't make calls between 2 sip communicator.. it mean i can able to make a call and receive.. but

hi all, i need to integrate and modify one of the application in asterisk/apps section... whenever i modified small steps ..in order to check and compile i 've to do recompile the whole asterisk module and it consuke to much time... please anyone couls you tell me, how can i modify it , compile and test the I/O in asterisk applications in a easy way... plz do reply .. Thanks for ur

Hi all, I need to connect two asterisk server in same network and i'm using sip user as my clients...... plz anyone suggest me.... Regards, Thiru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061227/aa4e409c/attachment.htm

hi, exten =>6000,1,dial(SIP/6000,15,tr) exten =>6002,1,dial(SIP/6002,15,tr) exten =>6004,1,dial(SIP/6004,15,tr) exten =>6006,1,dial(SIP/6006,15,tr) exten =>6008,1,chanspy(SIP/6006 | wbq) when i dial 6008 ,it is connected ,but i can't able to hear the voice of the any one. when coversation between the 6002 to 6006. in my Console mode i got the following comment *CLI>

In the past two days, I've been experimenting with the Pingtel SIP softphones and Asterisk, which one of my customers has been using. A few notes: 1) The "insecure=1" setting in SIP peers now works, from my limited experiments. Mark put this in for SIP servers that don't send requests in with a return port of 5060. This flag essentially takes any request inbound _to_ port

Hello All, I have pingtel and asterisk working really well. I have a really annoying little problem - mainly with pingtel. When a call comes in pingtel displays the caller ID on the phone. If I miss it then I click on the number for redial - this doesn't include a 9 to dial an outside line. The second problem is with the dialer from outlook again it bypasses the outlook dialing rules so

Hello all, Hope I am not too of topic here - but it cross's the phone/asterisk boundary. I have been playing with a few soft phones - noticed that pingtel seemed to be highly recommended across previous postings. I have been using xten - which is a great phone but seems a bit limited in its functionality - which is why I am now looking at pingtel. Problem is I cannot get it to

Hi! im trying tu run "microsoft office communicator 2005" and i cant resolve this: fixme:ntdll:NtConnectPort (0x1434f8,L"\\RPC Control\\epmapper",0x33ecd0,(nil),(nil),(nil),0x33ecf8,0x33ece0),stub! i google it all nigh long and i just cant find the way!!!. I need to connect to LCS 2005 because my company switch from Jabber to LCS. I tried pidgin and miranda-im+sip but didnt

Has anyone been able to get conference calls to work on the Pingtel Phones? I assume this feature works with their implementation, but connected to my asterisk box it doesn't work. The Pingtel phone thinks it is making a second call, but asterisk never sees anything about a second call. Any help would be appreciated. Sincerely, Andy Hester Consero

I just bought a Pingtel Xpressa from VoipSupply for use with Asterisk. I know that Pingtel has sold off their hardphone line and discontinued support for their phones, but I'd like to track down a few of the Java applications that they distributed before they went away, specifically their LDAP Phonebook app. Does anyone have a copy that they could send me? It was publicly

I downloaded the FC rpm for sip.communicator.org: http://www.sip-communicator.org/index.php/Main/Download yum localinstall of this rpm came up needing alsa-oss as a dependency. I have really tried to find this, looking at over rpmforge, kbsingh, and atrpms. So can anyone point me to this dependency?

Friday we'll be hearing about SIP Communicator Java VoIP and Instant Messaging client. SIP Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular VoIP and instant messaging protocols such as SIP, Jabber, AIM/ICQ, MSN, Yahoo! Messenger, Bonjour, IRC and a whole lot of other useful features. Open Source / Free Software, and is freely available

First I like to apologize if this is common knowledge, but I'm unable to get MSN messenger 4.6 to register with asterisk. I configured MSN messenger to use UDP and the IP of my asterisk server I edited the registry entry - for pC2PC calls under Windows98. What I'm I missing ? Asterisk version information Asterisk CVS-04/25/03-05:37:19 sip.conf [pingtel] type=friend

I have no idea what 'Yocto' is, but CentOS 7 includes two other tar utilities: 'bsdtar' and 'star' Maybe one of those will give you what you need? James Pearson ________________________________________ From: CentOS <centos-bounces at centos.org> on behalf of Klaus Kolle <klaus at kolle.dk> Sent: 16 September 2020 12:13 To: centos at centos.org Subject:

Hi All BT are providing a SIP gateway for PSTN through the BT communicator with Yahoo Messenger, I have done an ethereal trace and found that the BT Communicator side of the software is using SIP, so in theory I could add more PSTN lines to Asterisk for BT using SIP, but I am having problems deciphering the trace so my question is has anyone else tried to get BT Communicator work with

Hi. After successfully running ./configure I run make. When running make I get the following error.. [CC] ast_expr2f.c -> ast_expr2f.o [CC] ast_expr2.c -> ast_expr2.o [CC] strcompat.c -> strcompat.o [LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o strcompat.o -> aelparse aelparse.o(.text+0x3029): In function `ael_yylex':

Hello, I try to write parts of my code in C to accelerate the for-loops. But basic operations I want to do in R (e.g. start cluster). My R code looks something like this: library(Rmpi) mpi.spawn.Rslaves() mpi.remote.exec(....) dyn.load("test.so") erg <- .Call("test", ....) .... mpi.close.Rslaves() mpi.quit() And my C function looks something like this: #include

Has anyone have interfaced this successfully? I came to know from M$ that Genesys GETS can be used to interface asterisk. I have interfaced Cisco call manager to asterisk/ser but for my final setup I would like to have a LCS talking to a CCM, without having the Genesys GETS is I don't have to. Has anyone been playing around with this? If so I would really like to hear some advise.