similar to: Incoming Lines Confusion

Hi I want to create 2 groups of extensions, for example group 1 can't make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please

Hi, Yes InternalExtension is the context and 2093 the extension. Just to explain something odd that?s happening (and I?m very stumped with this) .I think my contexts are definately the reason that I can?t interrupt the menu for incoming pstn calls to choose a submenu: My users register with my sip proxy (SER). Therefore when I create an entry for them in sip.conf I set only one context. Also to

I am running Centos 6.4 64bit. Dovecot 2.0.9 I am getting the following messages in /var/log/secure, which looks like the pam authentication is not working but the users are allowed to login and the system works great. I am wondering if pam is actually failing and yet the system is getting the login info from elsewhere, or is this just a nuisance message? /var/log/secure Jun 12 23:11:29 smtp

All, I was a bit too focused on where I thought the problem was - turns out I wasn't crazy and the dialplan does work as expected. The problem was with dtmf detection - setting relaxdtmf=yes did the trick. Sorry for the premature post for help. Begin forwarded message: > From: Ben Witso <benw@bgwcomp.com> > Date: Mon Jun 21, 2004 7:28:42 PM US/Central > To: Asterisk-Users

Hi all, I am having difficulty getting incoming PSTN calls working. I have set up an account with a third party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and

I've got a telecommuter working out of her home office, using a Snom 200 phone, what happens is occassionally her phone will loose audio one way. She will be talking on a call that was incoming to her extension, and all of a sudden the caller can not hear her any longer, she can here the caller fine, this has happened with both Sip->Sip calls, and calls that have come in over our PSTN

How do you setup the timing in Meetme conference? I have a x100p and tdm4x card. When I dialing to my conference I get a request to schedule in the past error message. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 10, 2004 10:48 AM To:

Hi list, Currently, one of my phones registers fine, and the other does not, though for me they have the same config... Can somebody help debug/understand why? The logs in asterisk say: [Feb 24 13:48:09] NOTICE[20626]: chan_sip.c:15642 handle_request_register: Registration from 'IMSI208300618462231 <sip:IMSI20830061xxxx at 127.0.0.1>' failed for '127.0.0.1' - No matching

Hello, At http://syslinux.zytor.com/wiki is in the left hand side a menu called 'bits & bobs'. Please add "UseCases" and "SpinOff" to it. AFAIC see, requires it wiki admin priviledge. GSt

Gurus... If I set up my project to use SQLite (so it can come up) and attempt to connect to oracle via: oracle_connect = $application_properties[$connect_string] oracle_conn = OCI8.new(oracle_id, oracle_password, oracle_connect) where oracle_connect equals ''//bobs.big.boy.gov:1521/D09'' (with or w/o the introductory //) my rails app connect to the oracle database just fine. If

2013/8/13 Administrator TOOTAI <admin at tootai.net> > Le 13/08/2013 16:41, Olivier a ?crit : > >> Hello, >> > > Hi > > >> [...] >> >> >> How can I work around this ? >> Suggestions ? >> > > Answer the call, wait few seconds and then ring Bobs extension. If > asterisk detect fax it already sended to fax extension so

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change the default SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this

Greetings all: Newcomer to R as I work on learning it to transfer my college classroom stats training to something more useful and accurate then that spreadsheet from Redmond which shall remain nameless. I'm running v2.6.2 on a Win XP Home system that I keep up to date with all the called for patches. Haven't added much to the basic install other than the bits and bobs needed to run

Hi Daniel, I'm trying to add LivermoreLoops test to the benchmark suite (tar ball attached), but I'm getting the error below: --- Tested: 2 tests -- FAIL: SingleSource/Benchmarks/LivermoreLoops/lloops.compile_time (1 of 2) FAIL: SingleSource/Benchmarks/LivermoreLoops/lloops.execution_time (2 of 2) When I use the option to only run this test: --only-test

I recently began using the curl cmd to do an external callerid lookup on my own customer database. I've noticed certain lookups will cause a crash and not show anything in the messages file or the console. The curl command is connecting to an external webserver which has a oracle db connection. The file its hitting is PHP and does a very simply lookup showing the text like "C1234 Bobs

We have quadBRI configured 1 port in TE mode 2,3,4 ports in NE mode. We are trying to place a call from the phone connected to BRI card port #4 to city number through ISDN line connected to port #1. Number successfully dialed. Person on the other end answering the line. But conversation can't last more then 10 seconds. Below is a log of such call. Its not clear for me why we appear in

Hi All, I am using Asterisk for one of my projects in OpenBTS. I am having the age old problem of "extension not found" when try to make a call from one registered SIP phone to other registered SIP phone (two mobile phones connected to Asterisk via OpenBTS). The exact error thrown on Asterisk CLI is *"chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to

Hi all, -------- I have installed a TDM400 with one active FXS port (TDM10B) an connected it to a Siemens Euroset 2015 analogue phone. I have installed some smom IP phones to the network as well and configured them as usual (sip.conf). For configuring the TDM10B I have used FXO signalling in /etc/zaptel.conf and in /etc/asterisk/zapata.conf. I definded the TDM channel and the Snom phones to the

Ah, ok. I was wondering how you managed to build that, because when I tried it did not work, but I see you have backported also other bits and bobs. If those overwrite Base, perhaps add a warning of sorts. -- Sent from the Delta quadrant using Borg technology! Nux! www.nux.ro ----- Original Message ----- > From: "Richard Grainger" <grainger at gmail.com> > To: