similar to: Problems with ZAP dial timeout

First off, here is what I want to do: SIP Clients -> SER -> Asterisk -> VoIP provider Where SER will handle communications between SIP clients (since I would prefer that my SIP clients not use all of my bandwidth) Asterisk will handle calls to a VoIP provider I have read that people have similar setups working, but I have not seen any documentation of these setups. So far, SIP Clients

Sorry for the simple question. I'm trying to match "sipprovider.nocredit" but the following doesn't execute NoOp (it runs "context" but not "context-custom"). What am I doing wrong? [context] include => context-custom exten => _.,1,Set(GROUP()=1) exten => _.,n,Goto(destcontext,${EXTEN},1) [context-custom] exten => sipprovider.nocredit,1,NoOp(No

Hi, I am diverting an incoming call to a mobile phone and a landline using the following:- exten => 0203000000,3,Dial(SIP/442080000000 at sipprovider&SIP/44700000000 at sipprovider,120,r) For billing purposes, i need to be able to work out whether the diverted call was answered by the mobile or whether it was answered by the landline. The CDR only shows the full Dial() information, and

Hi, I have been trying to configure my Asterisk to use a Sip provider for out and incoming calls. I only have one user and password for connect to my sip provider. My sip.conf is: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No

Hello Asterisk users, Can someone thwack me with a clue stick please? I am following the Asterisk TFOT book Dial() example trying to get the limit and announcements to work as per below. These settings seem to have no effect. There are no warning messages after 4 minutes or every 30 secs thereafter and the call lasts longer than 5 minutes. gunner*CLI> show dialplan [ Context

Hi All, I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is james@james.com.... that is my username . Now... When I put this in the sip.conf file I have found that Asterisk is not able to parse it correctly and instantly goes to the email server to authenticate

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, it seems that there is a bug in asterisk's dtmf mode autodetection. Assume following sip.conf: [sipprovider] disallow=all allow=g726 dtmfmode=auto DTMF does not work. It seems rfc2833 mode is chosen despite it being obvious that this cannot work! The following configuration is necessary to get DTMF to work: dtmfmode=info In my opinion, this behaviour is counter-intuitive. I am using

OK, I need to dial a macro from AGI and needs to pass an argument. Ok, I found an bug report, but it was stated "un fixable?" really after 5 years? https://issues.asterisk.org/view.php?id=2470 I found this email in the archive, but no solution other then the dodgy work around? http://www.mail-archive.com/asterisk-users at lists.digium.com/msg85048.html I have

For some reason my voice mail stopped working properly. I was able to go in as a new user, set the password and options and now can never log back in using the password I assigned the mailbox. I can log in through the web interface with that password fine, and the voicemail.conf looks fine but every time I try to check messages I get "Password incorrect please try again" until it

I'm going to try and ask this again and keep it short and as too the point as I can while still providing enough info to be of use. PLEASE advise if I am going about this wrong or asking too much. I'm seriously doing my BEST to throughly read the docs and try a bunch of things BEFORE coming here to ask and possibly annoy. If is documentation that explains thsi process in terms that

I'm still trying to learn some parts of Asterisk, so sorry in advance for the dumb question! How do I set up an extension to dial out to the PSTN through my ZAP interfaces? I want the ability to have a ring group that will ring all of the phones in an office and then ring cell phones if nobody answers. I'm sure this is simple to do but I'm at a loss. I have tried the following

Problem 1 : Incoming conversations from the SIP-provider come into the [default]-context and to the 's'-extension. I am unable to change this, even if I have : sip.conf [general] ;context=default ; Default context for incoming calls register => 092779077:XXXX at 85.119.188.3 ; incoming [092779077] type=user host=85.119.188.3 context=from3starsnet So I define no

I'm trying to figure out how to allow an extension to register more than once. For instance, I have all of these 4 line IP phones that I use with Asterisk and I would like to have a persons extension (say 101) ring at all four lines so that if the person is on the phone they can take another call, but it appears as though if you try to register the same extension more than once then the most

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

Hi Guys, I am trying to make a web form where a person is allowed to put in $phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof caller ID. There are a few problems that I am facing with Asterisk AMI Originate command. The reason why I want to use the darn AMI Originate is because I am sending calls to mobile phones and I want to have some accountability and to know if a call was

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

Hi, I'm curious to know how I can programmatically differentiate between a Samba share and a Windows NT share from a Windows program. For both, GetVolumeInformation() claims the file system is "NTFS". This isn't very helpful. I need to be able to tell the two apart because "real" NTFS supports various things Samba's "simulated" NTFS doesn't, such as

hello I am running Samba 2.2.8 (latest) that I downloaded the binary from the Samba site. I have this problem in older versions of Samba as well. I have under Redhat 8.0 security level = user works fine under NT/2000/XP platforms, but when I try to login from a Windows 95/98/98se/Me machine, samba prompts me for a password to resource \\netbiosname\IPC$ how do I prevent this from happening in

I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots. It is annoying like nothing I can describe! I have tried Windows 2003