similar to: Broken calls

Hello Fellows! I have a TDM2400 and I can't put it to work. Every time it receive a call the Asterisk handle it and call the SIP phone; when people pick up the fone they don't hear nothing and the caller hear the phone rings and nothing happens. In Asterisk console I can see the message answered by the SIP's phone. I lost a lot of time trying to solve this problem without success :(.

I have a very serious problem with calls between PAP2-NA and a TDM2400 (8 FXO). Almost every call dropped after between 20 and 30 seconds with conversation. I disable the sound card, serial and other things on my server, but the problem still continues. I've changed the RPT Packet Size to .20 on PAP2-NA, but nothing. Here a piece of my log: [Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up

Hello All I have a TDM2400 card with 4 FXO, and the the following problem: This card answered all the calls, but for the caller, the call is ringing and I don't hear nothing when it has picked up. Here is piece of my log: Thanks for any help. == Starting post polarity CID detection on channel 21 -- Starting simple switch on 'Zap/21-1' -- Executing [s at entrada:1]

Hello fellows!!! I'm having problems with Ericsson Dialog 3221 phone and DISA. I've configured an extension to test DISA and it work properly with all other phones, but freeze with the mentioned phone. Here is my extension: exten => 105,1,Answer exten => 105,2,Set(TIMEOUT(digit)tting =5) exten => 105,3,Set(TIMEOUT(response)=10) exten =>

Hello All! I have a problem with my calls, that drops after 20 - 30 seconds. I got a piece of PAP2-NA log and Asterisk log and there's an error 603 - call declived, as showed. Thanks for any help. McCoy *********** PAP2-NA LOG *********** Feb 9 09:00:56 192.168.4.205 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060

Hey all, I have a Windows 2000 AD PDC that hosts a domain. He also trusts our existing Windows NT domain (2-way trust, they both trust each other). I also have a Gentoo Linux machine that I have compiled Samba 3.0 on. I can get almost everything to work with regards to talking to the Windows 2k PDC, like this: mccoy samba # wbinfo -u LIGHTSPEED+Administrator LIGHTSPEED+Guest

Hello! I want to get rtcp stats to cdr. (btw, I run asterisk 1.6.2.11) There is howto here http://www.voip-info.org/wiki/view/Asterisk+RTCP But I (and my users) do bridged calls from dahdi to sip, so in h extension channel is dahdi , and it doesn't contain rtcp stats. There is info about function shared. But I can't understand how to use it to put stats from sip channel to dahdi

Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition :   Video

Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not

Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing:

Hi all, I have a problem with some asterisk boxes. I have a standard installation with 1.4.14 (I also test with 1.4.4) in core duo Mac Mini on Debian Etch. I use SJphone softphone, Linksys SPA921 or Thomson 2030 for phones. All my phones are in a LAN with good status of 2ms max. Randomly I have dropped calls during communication. No absolutetimeout or other calling limitation options. Any

Asterisk Users, I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian "Etch" system. On the recorded voice mail messages, the volume is really low when retrieving them with my cell phone. I tried with multiple cell phones with the volume level high and still, the same problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it had no affect on

Hi list, Has anyone been able to get ISDN-BRI support to work reliably on Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro, kernel, modules, versions, config files). I've tried to get it to work on a Debian etch system with an HFC-PCI card and the zaptel package (v1.4.7, also from xorcom.com), but with no luck: all three channels that are created when the

Asterisk Project Security Advisory - AST-2017-012 Product Asterisk Summary Remote Crash Vulnerability in RTCP Stack Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity

Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machine....When running asterisk -rvvvvv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891

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

Hi list, Can someone please explain how to get one touch recording (automon) to work reliably? I'm using Asterisk 1.4.14 on a Debian etch system. My current configuration includes the following settings: In /etc/asterisk/sip.conf: [2000] ; Siemens Gigaset S675 IP wireless SIP phone. type=friend secret=1234 context=phones-j dtmfmode=rfc2833 qualify=yes

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

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

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 ??? --