similar to: Cisco 7960 and Asterisk...not working....

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

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

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

I am trying to setup a simple pstn gateway using Asterisk and a X100p card. I have got everything installed using Redhat 9 and am able to load Asterisk. I also configured sip and I am able to connect to the asterisk gateway with Xlite on the windows side. I am able to dial 1000 and get the welcome message. What I am NOT able to do is dial a seven digit local or 10 digit long distance number and

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

I'm able to receive but not make calls with zaptel using an X101P connecting to Asterisk with an Xlite client. My client has context = flat in sip.conf and extensions number 8919 In extensions.conf I've got: [home] ; Line 1 ; exten => 8919,1,Dial(${PHONES1},20,Ttm) exten => 8919,2,Macro(vmessage,${PHONES1VM}) exten => 8919,3,Hangup [outgoing] exten =>

Hello- I am in the process of adding a new provider to my asterisk box (both for outbound termination as well as inbound DID). They are going to be delivering and receiving traffic via SIP only. Now, in IAX via Voicepulse or others I know that I can simply have one registration statement along with an inbound context, then in extension.conf map the outbound context. from iax.conf: register

I am running asterisk 1.0.1 with OH323 compiled in. We have a 323 trunk to CallManager with a mgcp controlled pri router. When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using

Can you also post information such as: Type of phone (model Number would be idela) How is it conencted, SIP, ZAP, IAX, Channel Bank. Corresponding config files would also help. Help us help you. >>-----Original Message----- >>From: asterisk-users-bounces@lists.digium.com >>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >>Paul A Brown >>Sent:

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

You are reading the instructions for the STABLE 1.0 version of asterisk and are using the CVS version. Goto the wiki and read the instructions for RealTime. -Matthew ----- Original Message ----- From: "Muhammad Rizwan Khan" <rizwan@advcomm.net> To: <Asterisk-Dev@lists.digium.com> Sent: Wednesday, January 05, 2005 12:42 PM Subject: [Asterisk-Dev] Asterisk with MySQL >

Hi I'd like to allow users to transfer a call to a specific park number. This way, the receptionist can tranfer a call park for ext 100 at park number 7100 etc... It seems like this should be fairly simple using the Park(ext) app but it doesn't work for me. No matter what I extension I use, the system just picks the next available park number. I've simplified my dialplan for

I'm working on setting up a macro that will allow users to call their own DID number, and when they hear their voicemail greeting hit the * key and be prompted for their password to check vmail. For some reason though the background command isn't working as I'd expect it to: [macro-vmessage] exten => s,1,Answer exten =>

I am trying out a new service from www.talkn.com. They use Sipura to terminate your service like most providers. They are looking at directly connecting into asterisk in the future. Right now my configuration is Talkn?Sipura?Asterisk/FXO Card. When someone calls in and get?s answered and when either party hangs up , * releases the port with out a problem. If someone calls in and * answers the

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 =>

Hello All, After an incoming call goes to voicemail it doesn't hangup. Extensions Conf file [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip