similar to: SIP secret: argument only for outgoing

I tried putting this as the ata's dailplan: *St4-|#St4-|9|^9t4>$.- this is sip.conf [ata2001] type=friend username=ata2001 secret=SoMeSeCrEt host=dynamic context=fromata canreinvite=no and this in extensions.conf [fromata] ignorepat => 9 exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) exten =>

Hi, I'm trying to connect a PC with a TE410P to an E1/IP equipment. Unfortunately I keep getting a yellow alarm from zaptel (in zttool) and a Loss of Framing alarm on the remote equipment. The E1/IP is connected on the other side to a PRI interface on a GSM MSC. I have configured the span as: span=1,1,0,ccs,hdb3 (also tried span=1,0,0,ccs,hdb3) and the channels as: bchan=1-15,17-31

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: >

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 =>

On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp <jcolp at digium.com> wrote: > On Tue, Jul 9, 2019, at 7:00 AM, Dovid Bender wrote: > > Hi, > > > > Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you > > should be able to dial with SIP credentials in the DP. Is this still > > possible in recent versions of Asterisk either with chan_sip or

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

Hi All, I'm trying to connect to a SIP carrier who never connected with Asterisk. I managed to connect with a sipura phone or a grandstream, no problem. When I configure asterisk, I'm able to send out calls to the carrier no problems, however, receiving calls doesn't work, and I keep getting the following messages: <-- SIP read from 69.xx.xx.xx:5060: INVITE

Hello, has anyone a running setup for LMTP proxy and the director service ? pop3/imap/managesieve is properly working, but i have problems with LMTP. I set it up as described in the conf.d/10-director.conf From the user_db i get proxy=y and no proxyhost as described for imap/pop3 But lmtp is complaining about the missing host: Jul 07 15:00:48 auth: Debug: master in: PASS 1 user56

Okay, I am not understanding if I have this correct or not. I have a requirement to allow guests into a PBX from different domains. However, I can not allow the guests into the default context because each domain has its own IVR. So I end up setting the domain context. I also need to provide separate contexts for different sip users (different dial groups). Small system, few users, so it

I have configured yum to use a proxy via the yum.conf file by adding: proxy=http://myproxy.com:8080/ What I noticed when running yum check-up date is that some requests are going through the proxy while the system seems to be trying to resolve the domains of other hosts in the repo and trying to establish direct connections to them instead of going through the proxy. Can anyone explain this

Hi all, I have a IAX trunk between two asterisk servers, both with dynamic IP and both have a DNS name associated with it. In the iax.conf file I configure the "host" parameter with the DNS name of the servers. Everything works fine until one of these servers get a new IP, so the other can't find its peer (the one that has just gotten a new IP). If I manually issue a "iax2

Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss

For the record. -----Original Message----- From: ripley@stats.ox.ac.uk [mailto:ripley@stats.ox.ac.uk] Sent: Tuesday, June 18, 2002 3:21 PM To: Warnes, Gregory R Cc: 'r-help@stat.math.ethz.ch' Subject: RE: [R] Problems with url/download and http_proxy The port is not supposed to be required, so rather than fix the docs can anyone fix the problem? On Tue, 18 Jun 2002, Warnes, Gregory R

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

All I've suddenly lost incoming audio on my FWD connection. It worked fine up until Wed when all of the sudden my calls would complete but I couldn't hear any audio (I could see the status of the call on the CLI and could see that my call was using bandwidth on the ethernet switch and router). I swear I didn't change any of the configuration or even restart *, but all the sudden

Hi! Since 1.6, when using AEL, macros are implemented using Gosub(). Is there workaround to have MACRO_EXTEN also in this case? regards Klaus PS: I know I could use something like context fromSip { 11 => &myMacro(${EXTEN}) } macro myMacro(MACRO_EXTEN) { } but isn't there some workaround to achieve compatibility with 1.4?

You're right, thanks. However, now I ran into the next problem. This seems to be related to the fact that I have a 64-bit machine? $ ../../src/llvm-gcc/configure --prefix=/home/johan/llvm --program-prefix=llvm- --enable-llvm=/home/johan/llvm/obj/llvm --enable-languages=c,c++ ... $ gmake ... /home/johan/llvm/obj/llvm-gcc/./gcc/xgcc -B/home/johan/llvm/obj/llvm-gcc/./gcc/

Why will gems only install rails 1.0 when using a proxy? gem install rails -p http:\\myproxy will only install rails 1.0!!! why is this? it''s very irritating Chris -- Posted via http://www.ruby-forum.com/.

This does seem to fix my problem: > Sys.getenv("http_proxy") http_proxy "http://gproxy1.pfizer.com/" > url("http://cran.r-project.org/src/contrib/PACKAGES",'r') description "http://cran.r-project.org/src/contrib/PACKAGES" class

Hi again! About my previous E-Mail... I though about it and I think, that maybe I'm just very stupid... Since I called an INTERNAL number, Asterisk tried to call it. I tried right now to call an EXTERNAL number (using my context [myproxy]) and the behavior is NOT the same... Not 100% correct, but it tries the right way... Now my problem is to check in my dialplan if the peer, that