similar to: How to make RTP Packets NOT passing thru Asterisk?

Which platform? If on Solaris 10 sparc, GCC (either from Sun or sunfreeware.com) should be 64-bit by default. GCC from Sunfreeware for Solaris 10 x86 will compile 32-bit by default. For Solaris, you are better off using Sun Studio and Dmake. Actually, you are better off just using the compiled version from Oracle/Sun. On 06/13/12 02:08, prabu.murugan at emc.com wrote: > Hi, > >

Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI> rtp debug RTP Debugging Enabled -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered

Hi, This is Prabu. I am trying to compile samba on Solaris 10_U10. I have posted the same in bugzilla. https://bugzilla.samba.org/show_bug.cgi?id=8939 User requirement is to compile samba 3.4.17 to support their application. 3.4.17 is working on Solaris 9 and Solaris 10_U5. But not on Solaris 10_U10. ./configure and make gives error related to talloc. I have set the PKG_CONFIG_PATH. It is not

On 05/17/12 11:15, prabu.murugan at emc.com wrote: > > Hi, > > As a security concern we are upgrading samba to 3.4.17. > > I tried all possible option to compile samba 3.4.17 on Solaris 10_U10. > But it is not going through. > > > > User requirement is to compile samba 3.4.17 to support their > application. 3.4.17 is working on Solaris 9 and Solaris 10_U5. But

Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp

Dear all, I recently downloaded libvorbis from the Xiph.org website. I would like to compile it using a cross compiler because I would like to import it to the filesystem of an embedded machine. Hence, I tried to alter the CC of the makefile and change it to the directory of the cross compiler instead of the original gcc that was there. However, the change did not get implemented, and further to

can anyone tell what is the "unknown RTP codec 72" means and how to fix it. I'm using xlite to call PSTN line and the message just pop up on my console but the call can be connected. What am I going to do? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as

Hi, Sorry for my bad english but I?ll try to explain my problem I got an Asterisk running in my house with ADSL... I?m using X100P and TDM400P cards.... My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here?s the map with Firewalls Call for anyone to my house => PSTN => X100P => EXTENSIONS => SIP/RTP => ISA MICROSOFT

My first Asterisk install: Debian sarge with the 2.6 kernel, and two X-Lite soft-phones. I followed the online how-to documents and was calling between the two soft-phones and calling the demo system with no problems and had full audio. I then went on to configure the TDM400P's two FXS modules. I got into that a ways and was having some success, but no dial-tone when I was off the

Hello I thought I had things set OK to have Asterisk play FR files for prompts and MOH, but for some reason, it still can't find them: ============ ll /var/lib/asterisk/sounds/ drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/ drwxr-xr-x 10 root root 61440 2011-03-17 14:21 fr/ Note: fr/ contains core + extra + moh as downloaded from here:

I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent at best. Sometimes it's great other times completely unusable. When it's bad one usually hears harsh static when the other party speaks or their voice gets "clipped" to static if they speak too loudly. Many of these users have migrated to Skype ? much

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.

Dear all, I have tried to run an asterisk instance together with XLite on a single machine (a PowerBook). The intent is to take advantage of IAX connections to easily cross NATs while traveling. While the IAX setup proved 'easy', just having to fiddle a little with working configs at both sides, I did not succeed so far in getting XLite to connect to the local Asterisk server, AND be

Hi guys I have configured a samba4 ADDC. It working well but users homes share did not display. How can we solve this problem. Please help me.

Hello I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite on XP as an SIP client: http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png The problem is that by default, Asterisk doesn't rewrite the CID name + number in incoming calls, so that XLite displays whatever name I used in the 3102 and the extension the 3102 uses to register with Asterisk. How can I tell

Hi, I'm wondering if anyone has experienced an issue with the XLite softphone and asterisk accepting dtmf? I can listen to my voicemail perfectly from my hardphone. However when I dial the voicemail number from my XLite softphone and enter the password at the voicemail prompt, an error appears vm-incorrect and I get an "Unable to read password" message on the asterisk console. Has

> Sergio Gelato wrote; >> I see that the rest of that function has an "if (problem) goto out;" >> after >> every krb5 library call. Doesn't that also introduce measurable time >> differences? Interesting. > I wrote a test case with expect to measure the time difference for valid and invalid user with the same workaround as said before. It seems to

Dear All, I have downloaded the xlite version 2.0 for windows and I made the following conf in the xlite itself as the document suggested in order to make it work with Asterisk but still it doesn't work as a matter of fact when I tried to make a tcp dump I can see no packets going between the windows client and the Asterisk server at all, here is the my conf on the xlite itself: in the