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This is not an Asterisk specific question but doesn't anyone know if you can automatically prepend a 9 on the call lists so clients can return dial without having to repunch in the number? If you go to directories now it just shows the number without a 9 (obviously). Maybe on the Asterisk side?? Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

192.168.1.107 199 6bd3fb49505 00102/00000 ulaw No Tx: ACK 192.168.0.100 110 5c5a4953-65 00101/00005 ulaw Yes Rx: ACK Those channels are stuck talking to each other. The phones are disconnected yet that connection remains. I can clear w/ a restart obviously, but is there any way to tear down a call like that from the CLI? Bill -------------- next

I have a situation that has repeated itself a few times. Someone calls into Asterisk and is connected with a voice extension. At some point during the call, the log shows "chan_zap.c: DTMF digit: f on Zap/2-1". At this point, the call is redirected to receive a fax and the Asterisk voice extension is hung up. The users report that there were no noticable tones heard just before the

I have read that 1.4 has shared line appearances, which I assume will work with Polycom phones. Has anyone configured this and verified it working? I was going to start playing around with it but wanted to see if anyone else has tackled it yet. Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax server and having it then dial using iaxmodem is working fine. Hylafax server is talking to my Asterisk box that has a Sangoma A101 in it via iaxmodem via an IAX channel using ulaw. A call coming into a certain test DID comes into the Sangoma A101 then it goes to another box via IAX ulaw that uses the rxfax app to

2 Asterisk servers 1.2.12.1 Connected via IAX2, same switch, GigE, no packet loss, etc 1 with a Sangoma A101 for a PRI to the PSTN Ulaw QoS enabled NAT for the registered ATA boxes, no nat between the * servers Faxing inbound: Call from PRI hits the first Asterisk server Then talks to the 2nd via IAX2 NVFaxDetect receives the fax, converts to PDF and emails it out Works great!

I've been playing with Asterisk for a few weeks and it's working great. I have a question about getting multi-line receptionist phones working. I was thinking about getting one of these expansion ports: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0 9186a008008883d.html What are people using for receptionist phones that show all the extensions in

Hi all, I install the last version of Asterisk and I tried to send faxes, but nothing works. Here is my configuration: Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA <----> Analog Fax 2 I tried Analog Fax 2 -> Analog Fax but nothing works!! In the Patton configuration I put G711 and no silence suppression. In asterisk I have

...ID: <042220062334.2677.444ABD850006363400000A75220588617208010B020E9B02@comcast.net> Content-Type: text/plain; charset="us-ascii" disable three-way calling, restric channels to one per call. -------------- Original message -------------- From: "Bill Gibbs" <bgibbs@edurotech.com> > I say just bill the user at extension 333 it's his responsibility to > keep the login info private. If he disputes it, refund the first time > then change the password to something really complicated then start > billing him if it keeps happening after that! > &g...

Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? Thanks, Bryan Mahin Please visit us @

I have a Polycom 301, my only Polycom and it's been working great so far. The only thing I have noticed is that when I have 2 line keys with 2 different labels, it only shows the last "...XXX" where XXX are the last 3 characters of the description. If I have one entry, it shows the full name (like General or Billing). If I hit Menu button, Status, then Lines it shows the full

Has anyone had any luck getting listed in directory assistance when your number is ported from For example, I have an asterisk box for a client, that is also shared with another client in the same building. The CLEC provided PRI and numbers (including the ported #s from Verizon) as the main owner of the PRI (call them Tenant A) and the CLEC will not change the listing name for Tenant

I am having a problem getting the following logic to work, in a macro. Basically, if the caller ID matches, set the outbond trunk to a Zap channel, otherwise use a SIP provider. exten => s,n,Set(TRUNK=${IF($[${CALLERIDNUM} = 1234567890]?Zap/g1:SIP/LDPROVIDER)}) ; use PRI instead of SIP That works. The TRUNK variable is set properly. But the SIP LD provider requires a prepended

I have 4 Polycom phones with multiple line keys so multiple incoming calls work fine The way I would like the incoming call flow to work is as follows: 1) 2 groups consisting of 2 phones each 2) Incoming call rings the first group, if no answer, the 2nd group is rung 3) However if the first 2 are on a call or busy, it will immediately ring the 2nd group 4) If one

I am running some Polycom phones and have Auto Answer setup(*51 initiates that when you call an extension) With an attended transfer you can take a call, hit transfer, *51<extension>, announce the call and if the person wants it, complete the transfer, the call is now on speaker at the end. This can surprise people because all of a sudden the call is right there. I know that the

Any benefit on getting the PCI Express version? Bill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070222/62f79910/attachment.htm

Behavior on Asterisk 1.2.12, 1.2.15, 1.2.16 "sip show channels" Always tends to show 100+ lines such as 192.168.1.241 (None) 2e2872da-1d 00101/21507 unkn No Rx: REGISTER Never seem to go away 198 total peers on this server All devices are behind NAT Registration expirations between 30secs to 2 minutes to help keep NAT open Should I extend the

When you use the Park button, on some phones you have to hit "More" to get it. Then when you park it, it calls back and tells you the extension...so you have to hang up then pick up again. Callpark apparently is a valid extension! Bill -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Stephen Bosch

I have a PRI switch type national Asterisk 1.2.16 Zaptel 1.2.15 If I call an invalid number I get * PROGRESS with cause code 28 received Asterisk continues to attempt to connect the call until the timeout is reached and I hear ringing. I want to capture the progress code, which I thought was in HANGUPCAUSE but when I NoOp that variable it's always 16 when I dial an

Asterisk 1.2.16 I have an extension "102" with a Polycom 430 I am trying to protect against forwarding loops If I set the phone to forward the line to itself, extension 102 I get the following -- Got SIP response 302 "Moved Temporarily" back from 206.83.240.18 -- Now forwarding Local/102@mycontext-b2ee,2 to 'Local/102@mycontext' (thanks to