similar to: secret=pw in sip.conf affecting inter-asterisk sip call

I've just started consulting for a SME client based in Reston Virginia. They don't know it yet but they are going to need a hosted asterisk service and some DID's. Email me if you are able to provide 10 DID's in Reston (must be able to be ported away!!) and hosted Asterisk with end user configurable IVR etc. Probably only 5-8 users at the moment BUT... they'll be

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Hello. I'm new with asterisk. Can you help me in this: I have cisco sip phone (601) connected to asterisk server, and 1 client number (500). I want to dial from 601 to 500. But get error in cli console: [Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite: Call from '601' to extension '500' rejected because extension not found. What's wrong?

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension

Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '0426000000' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

I've setup Asterisk to work with Broadvoice for both incoming and outgoing calls. I can make outgoing calls, but when I try to receive an incoming call I see the following message on the console: [date] NOTICE[8661]: chan_sip.c:13178 handle_request_invite: Unable to create/find SIP channel for this INVITE It's registered with Broadvoice: Name/username Host Dyn

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

I receive an INVITE/SDP containing: m=audio 11310 RTP/AVP 3 0 101 which I interpret as gsm, ulaw, rfc2833. and I reply with an OK/SDP containing: m=audio 15884 RTP/AVP 0 3 101 which I interpret as ulaw, gsm, rfc2833. How can I tell which codec was actually used for the call? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards

Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk

I am up to post 5 on my step-by-step but I hit a bit of a snag and so far my searches have failed me, I hope someone can help. (By the way, I added an asterisk index for quick navigation on the blog http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html.) Here is the snag and I am hoping for a little help from the collective. Inbound I have 2 different numbers. I can call in on both

Hi All, I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet !

Hello, everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I get the following : [Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username mismatch, have <329909006666>, digest has <3291119600> [Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite: Failed to authenticate device "0473990000" <sip:0473990000 at

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf

Hi, I'm trying to get a softphone configured. In Sip.conf [general] I found an example that said I need: nat=yes localnet=192.168.xxx.xxx Is localnet supposed to be a LAN IP or a WAN IP? Thank you, Gary

Hi, One of our customers has an issue with the callee not being able to hear them. It seems to happen very frequently on one number in particular where there are about 3 IVR menus to dial through before getting to a live person. However, this does not happen on every call. Running tcpdump on the RTP packets, I can see that RTP is setting sent, but the values in the packet are all very close to

> On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: >> I receive an INVITE/SDP containing: >> >> m=audio 11310 RTP/AVP 3 0 101 >> >> which I interpret as gsm, ulaw, rfc2833. >> >> and I reply with an OK/SDP containing: >> >> m=audio 15884 RTP/AVP 0 3 101 >> >> which I interpret as ulaw, gsm, rfc2833. >>

Is anyone can help me with this ? I'm really desperate. Thx in ad. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ikka - Mitra Kreasindo Sent: Wednesday, September 14, 2011 5:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Mixmonitor command parameter problem on