similar to: RTCP too short

Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?: ? -- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack ??? --

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

Hello Asterisk list, I've facing a memory allocation issue that happens occasionally but on a consistent basis. The problem happens as follow, suddenly Asterisk starts consuming a lot of memory, in a rate of more than 1GB per hour. Kernel will eventually kill it via the OOM killer when memory is really exausted... This situation does not generate backtrace because Asterisk is responsive

I built asterisk-trunk at 92526 and asterisk-addons-trunk at 496. I have my Bluetooth cell phone connected. It almost works. In mobile.conf, I have "context=incoming-mobile" for the phone, and that looks like: context incoming-mobile { _. => { VoiceMail(9999,b); Hangup(); }; } Calls to the cell phone get directed answered by Asterisk and directed to

Hi, While looking for the cause of disturbance in call I found this error coming in console RTCP SR transmission error, rtcp halted Google search only shows some bug reports relating to MOH and Hold. What could cause this message? Could this be a symptom causing call disturbance? Where should I start digging to find out the reason for this error? I am using Asterisk 1.4.19 with zaptel 1.4.9.2

Running fc3 with current cvs-head... Is there a nice way to rotate the /var/log/asterisk/messages file without shutting down asterisk? I'm currently rotating the log files via cron, however my script requires asterisk to be shut down, which also kills any outstanding cli sessions (eg, asterisk -rvvvvv). Would like to rotate the files without killing the cli session. Any reasonable way to

Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126

Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why

Hi, I am plying with * for 10 days now. I am testing with a couple of vocaltec h.323 gateways (FXO and PRI) cisco ata-186 (configured for SIP) and MSN messenger (SIP). They all seem to interoperate. However I have a problem when * is sending calls to the vocaltec gateways. Vocaltec gateways are monitoring the RTCP reports send from the remote gateway (in this case *) and if they don't get a

Dear all; I want to stop RTCP from Asterisk-server to phone. But I want to use RTP. I looked rtp.conf/sip.conf, but I can't know about it. Please tell me how to stop RTCP only. Because , when I access under NAT, my gateway shutdown the port as gateway received RTCP from server. I use Asterisk 1.6.2.6 or 1.4.29 . Also SIP/RTP. thx.

This article below came up on the newwire. The RTCP XR RFC was published. Will Asterisk be supporting this function in a future release? Does anyone know if any phone vendors are going to be supporting it? Thanks Lee Goodman Our Technology Update this week is about one of those mechanisms. Known as RTP Control Protocol Reporting Extensions (RTCP XR), the technology defines a standard way to

Hi, It seems that asterisk doesn't send RTCP packets when a call is on hold. Is there any way to get asterisk to send these packets? I'm in the process of setting up a Lync (microsoft voice) server which will use an asterisk box as a gateway. The trunking between asterisk and lync is 'working' however when a call is put on hold asterisk stops sending RTCP packets to lync, and

(hello all, this is my first writing. so please bear with me if I'm wrong anywhere.) orry to break too lately, but how is the RTP payload submission is going? could we see the new payload at March IETF? I agree that it would be fairy straightforward to make an RTP payload for ogg vorbis, assuming raw packets, AFAIK. using physical bitstream is, in this case, not adequate by the reasons in

I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on this console is made up of weird characters instead of normal. Please see http://www.softins.co.uk/dsc00018.jpg for an example. If I do "asterisk -rvvvvv" on a normal login, either via the

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.

Hello, We are using Asterisk 1.4 as call distribution system with simple queues for SIP calls. With high load (4000 calls/hour) some calls remain in queue forever (until queue's max wait time) in spite of being hung up already by the caller. It seems that when a BYE is lost, Asterisk has no mechanism to check whether a call is still active. Is there a way to activate a RTCP call control,

I have asterisk 1.0.5 installed via apt on a debian system. It's a custom distrobution called Voyage Linux that runs from a flash card and I have a hard drive installed with mysql installed on it as well as apache. I have been though the AMP install guide (asterisk management portal) and in the interface it has a place for me to record new IVR menus. I have to dial *77 to begin recording

http://bugzilla.netfilter.org/show_bug.cgi?id=757 Summary: SIP connection helper not setting RTCP conntrack expectation Product: netfilter/iptables Version: linux-2.6.x Platform: i386 OS/Version: Ubuntu Status: NEW Severity: normal Priority: P5 Component: ip_conntrack

G'day. I am wanting to find out how my SIP service is performing with Asterisk, especially jitter and dropped packets. I can get an overview of that using the 'rtcp stats' function at the console, but is there any way to get those logged to a file or some other permanent record? Nothing in logger.conf seems applicable, save perhaps directing verbose messages somewhere, but it