similar to: Unable to connect to remote asterisk

Hi to everybody my system is be attack, but I dont know what this means [May 27 15:12:24] WARNING[26018] chan_skinny.c: Partial data received, waiting (76 bytes read of 786) [chan_skinny.c] skinny_session[0][C-00000000] skinny_session: WARNING[May 27 15:52:32] Asterisk 13.8.0 built by root @ asterisk on a x86_64 running Linux on 2016-04-04 19:02:51 UTC [May 27 15:52:32] NOTICE[2306] cdr.c: CDR

Hi List, The below error caused the Asterisk to crash, if anyone have idea on this please reply,(Asterisk version :1.8.9) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to

Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 <..> Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Hi again! I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI with Armbian/Debian 9. First test was to call a test service that say the time. Works! Second test was to record my voice and play it again. Works! Third test was to call the other VoIP-phone. It does NOT work... :( Then I noticed that, by starting, Asterisk says the following messages: [Feb 15 18:42:54]

Ok after I added tcp transport and disable force_rport phone get registered, but still have issues with calls, when I call from cisco from, it work except hangup. when I call to cisco phone asterisk return congested debug of call <--- Transmitting SIP request (952 bytes) to TCP:192.168.1.61:51179 ---> INVITE sip:111 at 192.168.1.61:51179;transport=tcp SIP/2.0 Via: SIP/2.0/TCP

Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten => _1X.,1,Dial(SIP/usman02/${EXTEN}) exten

Thank you. I read that last yesterday afternoon, and I could've sworn I tried that but I will look into it again (I've tried so many different things it was getting cloudy what I've tried and what worked etc, combined that the extension config gets messed up after playing with it so much so I'm often recreating it as well). I also found a bug report in the FreePBX bug tracker

I?ve gotten to the bottom of this; Seems that the pjsip.endpoint_custom.conf isn?t getting included properly, or my syntax is wrong. If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk, everything works perfectly. Unfortunately, I?m using FreePBX, so it owns this file and my changes won?t persist a FreePBX reload. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > Have you tried NAT=force_rport ? OK, tried... I can transmit from my phone (aka: I hear my voice on another phone), but I'm not able to receive data (aka: I cannot hear what I say on the other phone). Other suggestion? Thanks Luca Bertoncello (lucabert at lucabert.de)

HI all: I have spent a large amount of time configuring/installing phones connected to Asterisk. Halfway through the process I discovered that my Cisco7960 with 2 7914 expansions was not supported in the SIP protocol. After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of configuring SCCP to properly work with Asterisk. So far I have gotten the phone to dial and receive calls

Hi list, I've been googling this issue and found some good resources however I am still running into problems with the following combo ... Here's my story; - Asterisk 13.4 with FreePBX 12. - Migrating from Asterisk 11 / FreePBX 2.11 - Mix of Cisco 79xx handsets, mostly 7940G's. My problems started with (the very common) issue of the 7940 not replying to 401

Is there a digium twitter 'user' to follow that only tweets important announcements and release information? If there is not, I think there should be. It would be highly utilitarian to get an SMS when there is an update to Asterisk, Dahdi, ADA etc, but I don't want to be bothered real-time with asteriskpbx tweets like: "Anyone trying anything cool with Asterisk over the

Anyone in New Zealand using AsteriskPBX? If so, what hardware are you using to connection to Telecom's lines? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030619/d5c12ca0/attachment.htm

Hi, In digium website. http://store.yahoo.com/asteriskpbx/newitastdmde.html It is said Dev Kit PCI card works with 2.4 kernel. I am wondering if it is also working with 2.6 kernel? Anyone knows? Thanks Michael

Currently using PJSIP. First, they want me to get this working with the existing PJSIP configuration, but then setup a second box using chan_sip performing similar work. For PJSIP... I currently have an endpoint configured to a system using IP based authentication. It is configured with a match setting in the endpoint section. All channels coming from that IP address go to this endpoint. They

Thank you for the response Joshua . I had rtp_symmetric=yes before I wrote the email, then I set it to no, restart asterisk, and tried to make the call from the remote endpoint again but still tcpdump is showing me the RTP packets are being sent from Asterisk to the private IP. tcpdump on asterisk server showing UDP packet bound for my remote endpoints internal IP: 17:07:57.130212 IP