similar to: Delayed RTP

Hello, Has anyone encountered that when Asterisk sends RTCP messages, it stops sending RTP packets until it gets an answer? Can that be fixed? Thanks.

The strange thing is its only sometimes my dial string is as follows exten => s,1, Dial (SIP/200,, tT) For that particular route but obviously s,3 as have Ringing () first etc. After she pushes ## 6 times it will go thru sometimes. Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Kevin Larsen <kevin.larsen at pioneerballoon.com>

On connection to an incoming call via PSTN where asterisk [11.7.0] is Dialing an internal extension on answering the call there is about 6-7 seconds before audio is heard on either side. When looking at the CLI traces when I answer the incoming call that asterisk extensions were dialing, I see immediately upon answering >0xhexnumber -- Probation passed - setting RTP source address to

Hello all, One of the buildings I have an asterisk box deployed in is used by two small companies on two floors. They have an agreement between them whereby they'll answer each other's incoming calls and take messages if the office is empty / everyone is on the phone. Each of them has an ISDN BRI delivered to asterisk via zaphfc, then dropped into a context as follows: exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Hello, we're running a small Asterisk appliance on a PCengine APU2C4. Base operating system is FreeBSD 12-STABLE, most recent incarnation as of today. Since update of port net/asterisk16 to the latest bug fix revision 16.6.1, we face a severe "slowdown" of everything that the Asterisk core performs, i.e. outgoing calls are delayed ~

Here's one for you phone people.... An elderly lady phoned her telephone company to report that her telephone failed to ring when her friends called - and that on the few occasions when it did ring, her pet dog always moaned right before the phone rang. The telephone repairman proceeded to the scene, curious to see this psychic dog or senile elderly lady. He climbed a nearby telephone pole,

FYI, -------------- next part -------------- An embedded message was scrubbed... From: Colin Perkins <csp at csperkins.org> Subject: Working group last call: draft-ietf-avt-rtp-speex-05.txt Date: Mon, 31 Mar 2008 22:25:03 +0100 Size: 2054 Url: http://lists.xiph.org/pipermail/speex-dev/attachments/20080331/289739db/attachment.eml

Hello, I have a small LAN network where I am running a Jain-Sip softphone on two user pc's. These softphones are connected through Asterisk(Trixbox). Although the phones do work in providing an audio conversation, there is a long delay(about 20 seconds) in the initial RTP session setup. I have tried a few values for the buffer length including setting it to zero. I assumed this would

Hi Guys Is it possible to register Kamallio directly to our SIP provider then load balance the RTP to 2 asterisk servers? We cant do the registration from the asterisk boxes as we want to do it directly from Kamallio. Is this possible? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, Please find below an updated Vorbis-RTP Internet Draft document for review and discussion at the Xiph IRC meeting on Saturday. The changes in this version have been: Codebook caching mechanism Expanded SDP parameters Expanded MIME section Expanded introduction Packet loss section Minor tweaks and clarity changes to text There are probably some minor tweaks to the formatting needed

After just having update to the latest CVS I am getting the following message when I call VoicemailMain(): -- Executing VoiceMailMain("SIP/2302-1a12", "") in new stack -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Playing 'vm-youhave' (language 'en') -- Playing

Our setup: SIP IAX2 SIP <SIP Phone> <-----> <Asterisk1> <------> <Asterisk2> <-----> <PacWest SIP> | RTP \-----------> <PacWest RTP> I can make or receive calls from my SIP Phone, however voice only works in one

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

Hello All, Has anyone ever seen this before. This only happens when i'm on phone call -- Zap/2-1 is ringing -- SIP/2203-c48d is ringing -- SIP/2202-f2ad is ringing -- SIP/2204-11cd is ringing -- SIP/2205-ce62 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- SIP/2205-ce62 answered Zap/1-1 -- Hungup 'Zap/2-1' Jan

Hello, I am searching for a DTMF issue on my setup ( 2 years and counting ), and I am wondering why rtp.c has code to mute DTMF ( the rtp->dtmfmute variable ), but this same mechanism does not exist in dahdi. I am sending a DTMF over SIP w/ RTP & RFC2833 to the asterisk box with the dahdi card. The dahdi card sends it out on the PRI line. Trouble is, the DTMF is echoed back and the

Asterisk Project Security Advisory - AST-2017-005 Product Asterisk Summary Media takeover in RTP stack Nature of Advisory Unauthorized data disclosure Susceptibility Remote Unauthenticated Sessions Severity Critical

Hi All, Could somebody explain me how the timestamps are computed in asterisk while bridging two sip channels ? I've got situation with my provider, who changed some things in config and added some codecs (that much i know) and after that we got one way audio issues. It seems that the problem is with RTP timestamps. Within one outgoing stream the RTP timestamps are growing, as it should

My configuration is: [ISDNPRI] -- [CISCO AccessServer AS5350] --<H.323>-- [ASTERISK] -- [CISCO ip phone 12SP+/Skinny] When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN everything working ok (RTP is ok). But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone IP phone party can hear ISDN party, but ISDN (incoming) party canNOT hear IP phone party

lianghu xu wrote: > In a word, I don't what's the standard of speex payload format. > The file doc/rtp.txt is for what? Is it not for rtp payload? > I find that rtp.txt is more detail that draft02.txt > > Which rtp docment should be followed? > Anyone else has written the RTP wrapper already? Oh, I see. doc/rtp.txt was a very, very early draft. See the manual for a

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