similar to: fromuser & fromdomain

Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user762 at

Strange things are happening at my asterisk box :) I've got asterisk setup to dail out with sip to my SIP provider. I've got NO fromuser/fromdomain stuff setup in my sip.conf When I place a call with my Siemens Optipoint 400 SIP phone everything is allright, the From: header is stating the username of the Siemens phone. When I place a call with X-Lite the From: header is altered and reads

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 list, I have a problem when I host 2 SIP-accounts on the same Asterisk-server. Asterisk picks out the SIP-account on alphabetic order A --> Z. In my sip.conf : register => user1:passwd1 at server/user1 register => user2:passwd2 at server/user2 [YOCAN-3starsnet] type=peer host=server username=user1 secret=passwd1 fromuser=user1 accountcode=user1_in [ITCENTER-3starsnet] type=peer

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

I have an asterisk box registering 100 numbers on a voip provider. Numers are: 2546.1000 to 2546.1099 My problem is that every incoming call arrived to number 2546.1099 that is the last number to register on voip provider. The correct is call arrive in destination number. See this exaple: I call to 2546.1000. -- Executing Dial("SIP/25461099-08738060", "Zap/g1/3000") in new

Don't be afraid about the info that I'm going to post in this mail, but I want you to give as much info as possible. Also I want to show you what I've tried. What do I want When a voicemail-message is left via the Voicemail()-application, I want the .wav-file send to my mail-address as an attachment. My mail-setup I'm not using sendmail as MTA. I have msmtp as MTA and mutt as

Hello setting "nat=no" or omitting "nat=" in peer definition does not help either. Still no audio. Why do you think this is a NAT issue ? IP and port information in SDP-body is correct. Kind regards. On 12-08-16 09:25, ????? ?????? wrote: > > Try delete nat from 770000wrtc settings ice should do the same > > > On Aug 11, 2016 10:00 PM, "Jonas

First of all, if Asterisk is the client and it must register to the other side, does the peer\user entry have to be in sip.conf, or can it be in ARA? Second, why do all calls fall through to the last context specified, whether in that peer\user definition or not? I'm assuming it's a typo somewhere, but I can't find it. I had a full sip.conf, but axed a lot of the fluff trying to

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

Hi guys, I configure one Fedora Core Linux 5 for use with asterisk as gateway using Digium TE110P interconected in Alcantel 4100 I've set up it to register 100 voip numbers on my provider. All calls on Alcatel is send to asterisk. In some periods of day i receive this messages on asterisk console: Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

Hi, i'm having problems with broadvoice incoming calls. I can perfectly place calls but my Asterisk Box is having problems when registering with the SIP Proxy. Sometimes it register and the call gets into asterisk, but without sound (seems to be NAT problems) and sometimes its not possible for asterisk to receive the calls. Everything was working great exactly for a month, but a week ago it

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

Hello, all. Alas, another convoluted question. All the simple things are, well, simple so I suppose we only need to trouble the list with squirrely problems! We've noticed a call history problem when using Asterisk where the call history on the Snom phones (with which we are very pleased) reflects the number of the PBX extension used by the B2BUA to dial the end point. I assume the same

I'm a first time user of Asterisk and have a working setup which I find clumsy. How can I clean things up to make the dialplan easier to maintain? My problem ========== I have 6 public numbers that can reach 6 individual users. I have 6 lines like this in sip.conf: [general] register => 31307115622:secret@belcentrale-incoming/622 .... register =>

To Jonas: I have an asterisk box at home and I have this line in my rtp.conf file: rtpstart=10000 rtpend=10100 And My FW is setup to forward all incoming ports of range 10000-10100 to the asterisk PC. I've never had a problem since one year, but I have never received more than two simultaneous calls with SIP clients. Message: 5 Date: Fri, 13 Sep 2013 11:49:59 +0200 From: Jonas Kellens

Hello, I'm new to this list. I'm trying to configure my Asterisk to have user access their email. SO far users can leave voicemail but they can't access voicemail. As you can see I had sip.conf and extensions.conf below. Please advice how to access configure extensions.conf to have users access their voicemail. Thanks in advance. -motty SIP.CONF [general] context=default