similar to: rejected because extension not found in context 'introutingB'

Hello. Sorry to repost this message but, I don't have the original message in my inbox nor in my sent box. Well, last week I posted a problem I am having trying to use an asterisk server use a voip provider and a pstn. Pstn works fine but, I cant even connect to my provider's server. I don't know what I'm doing wrong. I tried using a soft phone and I'm able to register and

Hi everyone, I have a number of SIP registrations going fine, but am trying to get a new provider going, and they have no sample Asterisk SIP config. They have been helpful, but keep falling back to the way they "think" packets should be flowing, and I've been trying to figure out how the Asterisk config should look like to get the SIP packet to look correct. Now, they say that

Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

Can someone tell me what I am doing wrong? Why doesn't CDR(duration) or CDR(billsec) return the correct values? cdr.conf endbeforehexten=yes extensions.conf [macro-Dial] ; ${ARG1} - Dial String exten => s,1,Dial(${ARG1},,M(post-dial)) exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long, billed for ${CDR(billsec)} seconds) The log shows: -- Executing [h

Evening everyone (obviously depends on when you're readin this, but hey). I'm trying to set up a multi * server situation, and am falling over at the second server, and after a day of google etc, have come up against somewhat of a brick wall. I can make calls each way between the two servers no problem, and can include the required extension at the remote * server as part of my main

Hi, I am attempting to make about ten calls simultaneously and intermittently get 'SIP/voipprovider is circuit-busy' followed by 'everyone is busy/congested at this time" I am not sure if this is related to my bandwidth to my voip provider, a configuration issue or something else. Has anyone seen this before and have any suggestions. Thanks in advance. --------------

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the

Hi all, The firmware I used is pa168s_iax2_us_151011.bin. My problem is the handset dial before I finished key in all the numbers, no matter how fast I managed to press the keys. It appeared it always dialed immediately, for example "011862", when I actually ment to dial 0118620xxxxxxxx. Thus left the remaining numbers "0xxxxxxxx" unsent. The handset had its dial plan

Hi list, My Asterisk config for outgoing calls is the following: exten => s,1,Dial(SIP/${MACRO_EXTEN}@voipprovider,60,g) exten => s,n,GotoIf($[\"${ANSWEREDTIME}\" = \"\"]?pri:hang) exten => s,n(pri),NoOp(Problems with voip provider trying PRI) exten => s,n,Dial(Zap/g2/${MACRO_EXTEN},60,g) exten => s,n(hang),HangUp in most cases it works well but, if my

Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme

Hello. >> ubuntu*CLI> sip show peers >> Name/username Host Dyn Nat ACL Port Status >> voipprovider xxx.xxx.xxx.xxx A 5060 Unmonitored I've ben trying to connect an asterisk server to a voip provider, and I'm currently wondering what the 'A' in the ACL field of the 'sip show peers' command might

Piling on... InterNIC says the domain was created almost a week ago, and expires in a year. The registrar is GoDaddy. The owner of the site is located in the Dominican Republic: C/1ra #15 Costa Criolla, Km9 Carr. Sanchez Santo Domingo, New York 00000 Dominican Republic Registered through: GoDaddy.com, Inc. (http://www.godaddy.com) Domain Name: CDSPORTAL.NET Created on: 14-Mar-08 Expires on:

Hello list ! I'm having one way audio on incoming and outgoing calls. Outgoing audio works fine, incoming audio is not working. My setup is the following : incoming calls : PSTN -- FXOport -- HT503 -- WANport -- Asterisk -- WANport -- HT503 (the same) -- FXSport -- DECTphone outgoing calls : DECTphone -- FXSport -- HT503 -- WAN-port -- Asterisk -- internet (VoIPprovider) I've done a

If you asked this question on the biz list you would get a lot of people that will tell you that they offer services where you can set the caller ID to what ever you want. To name a few:: Nufone Teliax Voipjet ----- Original Message ----- From: "Doug Crompton" <doug@crompton.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Unfortunately, I've ran into a second issue where addpd is being performed on memory that isn't 16 byte aligned. Again, this only happens if the createJIT OptLevel is set to Default (vs None). According to

Hi, I would like to access the AsmPrinter MachineFunctionPass during compilation in order to do alternative dumping of instructions, instead of using MI->dump(), which can get a bit messy. Is there any way to access this object or the assembler strings? It seems that even these strings / methods are not available through any static methods. Does it exist during the whole compilation or is it

On Wed, Jul 17, 2013 at 6:48 PM, Rafael Espíndola < rafael.espindola at gmail.com> wrote: > > Hi Rafael, > > > > Did this discussion ever get a conclusion? I support enabling > > pipefail. Fallout for out of tree users should be easy to fix. As we > > learned from LLVM tests, almost all tests that start to fail actually > > indicate a real problem that

Hi, I have 2 g729 licences - what I want to do is use g729 by default but if I get more than 2 calls at a time, use gsm for the others. So, I put this on all my sip providers: disallow=all allow=g729 allow=gsm However, this just seems to use gsm for everything. If I comment out the gsm line, it then uses g729. I thought it would use the codec's in the order they are allowed - is this