similar to: Record() cut off after 40 sec

Hi Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account. www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations. Many Thanks Scott

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

TL;DR: It's okay to close a bug, if you can justify it properly. Recently there has been a spate of bug-closing with what I would call inadequate documentation. Comments such as "Obsolete?" or "I assume it's fixed" could be applied to nearly every open bug we have. While this does reduce the open bug count--something I have been watching with morbid fascination

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

Isn't svn set up to auto-parse and post to the bug so you can just say "fixes bug 44444" and it parses it out? I mean, i added that to gcc like 15 years ago, i'm surprised we don't do this :) Nobody should have to add this info manually unless someone forgot to put it in a commit message. On Tue, Jun 12, 2018 at 1:36 PM, Tom Stellard via llvm-dev < llvm-dev at

I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[password]@voiptalk.org/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP

https://gcc.gnu.org/viewvc/gcc/hooks/ is how it was done. This used the incoming email handling for bugzilla i set up. These days, you could just use bugzilla's rest API IE a simple variant of https://github.com/mozilla/github-bugzilla-pr-linker/blob/master/app/app.py should work as a commit hook. That thing is written as a service, you just need the find/add parts of the rest api, rip

Why is it that newcomers always feel like inserting 'Answer' is a necessary step in their extension.conf entries? >[voiptalk.org] >;forwards any calls starting with an "8" thru voiptalk.org >exten => _8.,1,Answer >exten => _8.,3,SetCIDNum(55555555) >exten => _8.,4,SetCIDName(My Name And Surname) >exten => _8.,5,Dial(SIP/${EXTEN:1}@voiptalk.org,,g)

I've installed successfully freePBX with Asterisk, and got various internal extensions working, however. recently my internet facing IP address has been removed by my ISP (for various reason) and I'm not going to be able to get it back for a few weeks. Is there anyway in which I can successfully receive incoming calls from my Voip-Talk.org numbers (an 0845 number) without the static

Trying now to set up the final part of my * switch. I must admit I've had great fun over the last week or so playing with it, and would like to thank you guys for all the assistance (past and present, since I've been trawling a lot of old posts!!!). Scenario - using voiptalk.org to supply the incoming gateway, tied to an 0845 number for convenience in testing. Internal 7960 -> 7960

Hi, In current (6140c3177a0330f42411618c3fca28930ea02a21) samba's ctdb/tools/statd_callout_helper I find this comment: notify) ... # we need these settings to make sure that no tcp connections survive # across a very fast failover/failback #echo 10 > /proc/sys/net/ipv4/tcp_fin_timeout #echo 0 > /proc/sys/net/ipv4/tcp_max_tw_buckets #echo 0 >

hi, all asterisk 1.4.24 , zaptel 1.4.10.1 , E1 Manager API Action : Action: Originate Channel: ZAP/G1/8888888 Callerid: 12345678 Context: callout Exten: s Priority: 1 extensions.conf [callout] exten => s,1,Answer() exten => s,n,Wait(10) exten => s,n,Hangup() when the phone 8888888 pick up , it will come to callout context, after hangup, one cdr generate, but the

I have several of incoming numbers on IAX from voiptalk and magrathea but have a problem with IAX.conf. If I follow the example from voiptalk [VoIPTalk Incoming Number] type=friend username=VoIPTalk Incoming Number context=[XXXXXXXX] and make incoming entries in IAX.conf for the numbers like below with a different entry for each number pointing to a different context, incoming numbers always

Hi All, I have just installed Asterisk and am trying to connect it to a SIP account that I currently have with www.voiptalk.org but without any success. Although I know that voiptalk do provide asterisk accounts I don't want to convert the SIP account until am happy that it's gonna work for me. The asterisk box is currently behind a firewall and the following ports are being forwarded

Hi, I hit a bizare problem with dovecot 2.2.7 on debian 9 with LMTP enabled and auth/penalty disabled as documented here : https://wiki.dovecot.org/Authentication/Penalty Use case : I run a swaks command to send an email to an exim4 that tries to make a callout to dovecot-lmtp. At RCPT TO: swaks hangs 5.0<something-small> seconds then process normally (exim is waiting for callout

Is it possible to rsync through a firewall that requires HTTP basic authentication? The RSYNC_PROXY variable seems to correctly direct the request to go through the HTTP proxy server on the firewall, but there's no way to specify a username/password combo. The error message reported by rsync is "bad response from proxy - HTTP/1.1 401 Authentication required", which is not

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Hi All, I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan: [callback-outbound] exten => _00.,1,Macro(callout|${EXTEN}) [macro-callout] exten => s,1,AGI(getchannel.php|${ARG1}) exten => s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr) exten => s,3,Hangup() but for some reason i am not receiving the argument: Executing [s at macro-callout:2]

Hi all, Does anyone know of any providers that can offer local numbers based in the UK via IAX or SIP? We're looking at getting a number based in the UK. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

I'm not sure if this is a -users or a -dev question, since the answer probably comes down to something in the code. I'm running the latest CVS-STABLE, and am subscribed to PSTN service using IAX2 via Voiptalk in the UK. I've just been alerted by a customer that the sending of DTMF from my asterisk box to a remote PSTN user doesn't work, although it used to. To test it, I have