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