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When I want to pick up a ringing line, I dial *8 and hit New Call softkey on my Poly 501. For some reason, if I pick up the hand set and dial *8, it seems to ignore or drop the 8 digit. I've confirmed that this happens with all of my 12 Polycom 501s. Does anyone know what would cause this or how to fix it? Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on

Clearly that's not the answer I'm looking for. I need a hardware phone for this person. As the following post points out, a software solution is less than ideal for many reasons. Not the least of which is my user does not have a PC (please God, don't suggest that I buy a PC). I need a phone with a busy lamp field. Any suggestions? I'm leaning towards the 7960 + 7914 but I was

I'm attempting to set up call recording with Asterisk. Using automon => *1 ; One Touch Record in features.conf does not appear to be working. I'm using Polycom 501's but when someone dials *1 while in a call, nothing happens. I'm wondering if the phone or Asterisk is even detecting the DTMF. I suspect that is the problem but don't know how to verify or

My carrier tells me our Adtran is seeing error seconds and timing slips. Is there any way to check this on the T100P, maybe in /proc? Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100

We use Polycom 501s here and several users utilize the Forward soft-button to forward their extension to another extension or outside to a cell phone when they are out. My question is, how can I configure the dial plan so that if they have forwarded their extension via the phone, and the extension they forwarded to does not answer, return them to the voicemail of the originally dialed extension.

My 7960's seem to be stuck requesting CTLSEP00036B75B542.tlv from TFTP. I tried touching that file, but it just keeps requesting it. The phone is using SCCP 7.2. Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100

Are there any known issues with Cisco 7960s and any particular TFTP daemons? I cannot seem to get mine to speak to a Linux box, but Solarwinds under Windows works like a charm. Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100

I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is possible to customize the email message sent to the pager email address. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 agoss@ntad.com

I'm still having problems getting this to work. I cannot get anything to display on my 7914 other than blank lines. I have SIP/5920-5930 in [main] that I'd like to add to the 7914 and indicate hook status. The 7960 is registering okay as SCCP/5000. What exactly should my sccp.conf file look like? When I make changes to this, how do I enact them? Do I reload Asterisk and reboot the phone

I'm attempting to set up my SoundPoint 501 with my Asterisk server. I've configured DHCP and TFTP and successfully updated both the BootRom and SIP application. I've also created a custom cfg file for this phone's MAC address and the settings seem to be taking just fine. I can see that the phone registers with my Asterisk server but 'sip show peers' reports that the phone

If anyone could tell me what this error is all about, I would be very grateful. Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not permitted Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not

Whenever we call IBM, the call counter on the phone never starts and in the CLI the zap channel never gets the answered signal from the PRI. See below. -- Executing Dial("SIP/5933-645d", "Zap/g1/18004267378") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/18004267378 At this point, I am in IBM's menu system. However the call never

Does anyone have a recommendation for a good SIP phone with a busy lamp field? I need my operator to be able to see extension status for about 20 extensions and transfer via HOLD + extension button. I've got a pair of SNOM 360s with the sidecar, but I'm very disappointed with them. The buttons are cheap and rubbery like a Sipura 841, the handset cord is short and cheap, the audio quality

> Watch the output of 'pri debug span 1' on the Asterisk server while > placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) > might be relevant. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

Thanks for the reply. Forgive me for being na?ve, however have jumped in to this asterisk project at work due to some circumstances beyond my control and I don't know a lot about carriers and how this all works. I am figuring it out, but it's a lot of trial by fire. As far as I know, we only use 1 carrier for our system. We have a PRI from NuVox and we use 7 channels for our asterisk

I can not figure out what the heck is going on. I went back to my old version and I still get errors when the voicemail system tries to load any of the greetings, unavail messages, etc. the normal voicemail prompts work, but any user recording don't work. Leaving a new message appears to work, but the system wont replay them, it throws errors. Here is an example of the errors: Oct 11

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216