search for: fellipe

Displaying 7 results from an estimated 7 matches for "fellipe".

Did you mean: felipe
2011 Jun 28
1
Asked to transmit frame type slin, while native formats is 0x8 (alaw)
Asterisk 1.8.3.2 I have been getting this warning constantly on CLI in a call scenario where I use local channels to connect SIP with PSTN. I use callfile and local channel to first call a PSTN number and if answered, use local channel to call SIP phone with music on hold enabled in Dial string. If I call PSTN from SIP directly or vice versa I don't see this warning coming. On SIP I have
2011 Mar 15
2
Some errors
...t reached on transmission 521df80947598d560109d73f1493a76d at 172.16.1.127:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. Packet timed out after 32000ms with no response Well... everything works fine, but I don't like this errors, any ideas? Thanks for all. Best regards, Fellipe -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110315/6c74850a/attachment.htm>
2015 Jan 20
1
[PATCH] Makefile: add support for git svn clones
Fellipe, CXXR development has moved to github, and we haven't fixed up the build for using git yet. Could you send a pull request with your change to the repo at https://github.com/cxxr-devel/cxxr/? Also, this patch may be useful for pqR too. https://github.com/radfordneal/pqR Thanks On Mon, Jan 1...
2011 Feb 14
1
Problems with realtime sip
I have a problem using asterisk 1.6 with realtime sip. When I add sip channel (my sip provider) to asterisk using realtime sip (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip), incoming calls don't work for me. In asterisk CLI I get message: NOTICE[19805]: chan_sip.c:21250 handle_request_invite: Sending fake auth rejection for device "test" <sip:test at
2011 Feb 21
0
SIP METHOD BYE
...WARNING[25204]: chan_sip.c:3621 __sip_autodestruct: Autodestruct on dialog 'ee162385cac5cc9c at 10.1.1.13' with owner in place (Method: BYE) All inbound calls are fine. In other SIP users everything seems fine and I can make outbound calls. Asterisk: 1.8.2/ Any idea??? Best regards, Fellipe -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110221/66258346/attachment.htm>
2011 Apr 04
2
WARNING chan_sip.c:3115 __sip_xmit
Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",
2015 Jan 19
3
[PATCH] Makefile: add support for git svn clones
On 19/01/2015 4:13 PM, Nathan Kurz wrote: > On Mon, Jan 19, 2015 at 1:00 PM, Felipe Balbi <balbi at kernel.org> wrote: >> I just thought that such a small patch which causes no visible change to >> SVN users and allow for git users to build R would be acceptable, but if >> it isn't, that's fine too. > > Felipe --- > > It would appear that you are