similar to: X-Lite and Asterisk RTP cutting out

I have an asterisk server up and running, using Firefly in IAX mode works great, even with Firefly behind a NAT (as expected, since IAX works really well with NAT). Now I'm trying to get X-Lite and/or Firefly to work in SIP mode from behind the NAT, and I can't seem to get there. At this point, the phone will successfully register with Asterisk, and the Asterisk qualify messages get

I'm playing with codecs at the moment and have found some notices errors when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was getting garbled sound, but after changing magic number for both codecs to 97 (as per http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT,

Dear All, I installed an Asterisk on a linux PC, and X-Lite on two Windows PCs, all in a LAN. But, when I make phone call from one X-Lite to another, I always get Call Failed: 404 not found. Here is my sip.conf: [Phone1] type=friend host=dynamic ;defaultip=192.168.1.103

I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa (ATA -----> Asterisk ----> SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP

Hi there, i did some tests with two Cisco 7912G phones (SIP stack) yesterday. With both ethereal and tcpdump listening on the Asterisk-Server's NIC, it came up that all RTP packets were doubled, with some small but almost constant delay (~460 us). The setup is 7912G <--> ASTERISK <--> 7912G The tcpdump output shows RTP traffic ASTERISK --> 7912G: 000000 IP $ASTERISK.17944

On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp <jcolp at digium.com> wrote: > On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote: > > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ > > This specific issue exists in a lot of different implementations and > devices. Unfortunately there's nothing within SDP that guarantees or > provides what the source of

Hi all, I get "Unknown RTP codec 72 received" message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN over voicepulse connect (IAX) and to FWD echo test (SIP). But this message only with one SIP client, others (X-Lite too) not giving this message. All X-Lite settings are identical. Asterisk is last cvs version This what I see in console

sorry to bother with this trivial issue, but i am loosing all my hair ;-) got 2 x100p's and * on a slakware box x-lite to x-lite works fine! i also have: #ztcfg -vvv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. and in extensions.conf i got: [locals] exten

Hello When I make calls from home to the PSTN by going through the Net -> Asterisk -> the Net -> VoIP provider -> PSTN, I get no sound either way. I assume it's because I must tell Asterisk to use fixed ranges of UDP ports and map ports accordingly on the NAT firewall under which it is located on the LAN at work. Here's the schema: home > NAT > Internet > NAT

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

Hi! I decided to apply Chris's patch for the rtp problem, it is working just fine now. Thanks Chris!. I think that Mark should submit it to the CVS. Ildefonso. icamarg@unet.edu.ve >Pete, > >Try this patch below... I noticed that eStara's softphone has the same >problem as xten's softphone when it comes to DTMF. Seems as though = >Asterisk >is not looking for

Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite

could anyone help explaining what does this error mean? i get this error when make a video/ audio call from X-lite to Bria prof. phone rtp.c:1739 ast_rtp_read: Unknown RTP codec 126 received from '10.0.0.26' Gres -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, i'd like to know more about this issue, i'm always getting this message while in call with anyone from sip to zap or zap to sip. ast_rtp_read: Unknown RTP codec 72 received here is my current setup: client side, x-lite, with the transmit silence to yes, using ulaw,alaw on asterisk server side: sip.conf contain allow=ulaw and allow=alaw dtmfmode=inband So i always get this anoying

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing

Hi, I have the following config: [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> [X-lite] Is there a chance to avoid the RTP stream from passing through the Cisco CCM ? I would like to have all RTP handled by the *. This is just a testbed, for a larger project. What I want to achieve, is actually this: [Cisco Phone] <--skinny--> [Cisco CCM]

Not sure what a payload of 75 bytes could be. Are you sure that doesn't include the 12-byte RTP header? Other than that, maybe it's using wideband (although there's nothing that corresponds exactly to 75 bytes. If you send me the hexdump for one packet, I may be able to find out what it is. Jean-Marc Pawel Cyrta wrote: > Hello to all speex developers, > > I have question

Hi all, I'm working on an implementation of VoIP en Linux. I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a Red Hat 9.0 (*.*.*.172) with another softphone X-lite. Both of the softphones are registering and appear in the peers (sip show peers) with the good parameters of address and port. If I try to make a call, * receive the INVITE request and send a 404 NOT FOUND answer.

Hi, (moving back to the list as some bits can be useful to everyone) > I am sure that 75 is the length of payload only. Confirmed > I also don't have idea for the package containing data of a length 46. > > To make everything more clear, Payload type in rtp packages is 97. > SDP defines the stream as > a=rtpmap: 97 SPEEX/8000 That's a dynamic payload number, so it

Hello to all speex developers, I have question regarding payload length of narrowband speex in RTP. I were watching tcpdump of the xlite softphone and have found that it uses weird payload length namely 75 Bytes I went through various source and without success. To be clear: For 8000Hz sample in 20 ms that is 160 samples per frame. This makes 50 frames per sec. modes bit-rate 8 kbit/s