similar to: Different rings

I'm installing my first Asterisk server. I have a TDM04B card installed in my asterisk server (4x FXO ports). I have 5 Cisco IP phone 7960 working fine on asterisk using SIP. My configuration to receive call is working as expected meaning anyone calling on one of the 4 FXO ports is answer by asterisk and asked to enter the extension of the person to reach and then it is transfer on the

I'm wondering if there's a way it will show on the phone when there's a new message. Here's what I'm trying to do : in my extensions.conf when someone call from a PSTN line on my TDM04B card they have a choice. When someone press 1 for sales then I have 3 phones ringing at the same time. Each phone as already there own mailbox because if someone know there extension

Ok I know I'm not the only one having echo problem with asterisk but the weird thing is that when I receive a call from a PSTN line on my TDM04B card I don't have any echo problem at the beginning of the call then after a few minutes I start having echo on my side only (the person calling from a regular phone doesn't have any echo), then it stop and come back all the way until the

Abhishek, In reverse order 3/ yes it is freeware, though some of the termination boards are available for sale from www.digium.com 2/ yes you can interface to Cisco handsets running SIP. 1/ Does it have a gui interface - the short answer is no. The longer answer is depending on what you mean, if you mean programming - then no though a number of people have developed sql interfaces. If you

Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten => i,1,Playback(ss-noservice,noanswer) Exten => i,2,Congestion(15) Exten => i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice

Hi! I am having difficultly in having users of various SIP devices obtain the correct behaviour when they call a busy number ie. only hearing the Congestion/Busy tone. I assume this might be because the SIP device itself generates the 'ring' tone? With my current setup in the dialplan extract (below) the user of the SIP device hears one 'ring' and then the busy tone if a number

Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it work fine too. The problem is when a client calls an Asterisk extension, and Asterisk transfers

I have a small home setup with one Cisco 7960 SIP phone, one WISIP, one FXO connected to Bell Canada PSTN, and four FXS connected to POTS phones throughout the house. I also have an account to a SIP based DID provider. My problem is when I'm on a call on one of the FXS connected phones and receive another call either via the PSTN line (assuming the call I'm on is using my SIP

Hello All, We recently changed our asterisk system to begin using G.729a as the primary codec. We have a Cisco 1700-series router which connects to the PSTN via FXO ports, along with Cisco 7940 SIP phones. Everything is working great, except... When an inbound caller calls into our system, they hear an IVR. When the caller dials an ext (SIP phone), the ringing progress tone is

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about

Just installed the new 4-port FXO card and moved two pstn lines from existing x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified

Hello folks. I'm not sure if this is the right list for this question, but I'll start here. If I'm using a SIP provider and I have an entry in sip.conf that looks like: [8315551212] type => friend ... dtmfmode => inband ... When I pick up the phone, call someone through this provider, and press numeric digits to generate dtmf tones, who is actually generating the tones at the

Hello Im having strange asterisk ring. I'm dialing PSTN network, then I get my call answered and I hear a person talking but the same time remote person can't hear me. They get a ring tone. What can be the problem? Where do I need to look for it? have no clue. Running Asterisk 1.2.14 svn rev 48468 Voip gateway is Cisco5300 with IOS 12.3(9) Scheme: 192.168.1.201

I see that there's now a 2.16.1 upgrade path for Cisco ATA-186 devices, dated (variously) July 11 or July 14 2003. Here are some interesting bugs that claim to be fixed. Most notable is CSCeb17953, at least from my perspective, as I've hit this bug before. CSCea42480 The Cisco ATA ignores the Require:100rel header and processes call. CSCea69889 The Cisco ATA builds a 302 Moved

Anyone using the recently MAC OS X ? Version of asterisk ? Thanks, Francisco Perez-Landaeta > From: asterisk-users-request@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > Date: Fri, 27 Aug 2004 13:08:24 -0500 (CDT) > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users Digest, Vol 1, Issue 5082 > > Send Asterisk-Users mailing list submissions to >

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

I already have one Digium TDM04B installed in my server working fine. I just received 2 more so I did added them to my server. When I booted again linux told me that it found new hardware I said ignore. Then I log in as usual. I did ztcfg -vv to see if it sees 12 channels now instead of 4 but it only see 4. So then I did again modprobe zaptel, then modprobe wcfxo and wcfxs hoping it would see

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y So far so good. But I want to setup VOIP sessions with local carrier. I added dial-peer 40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN

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