similar to: Call Recording with *

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

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

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

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

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

For whatever reason, the music on hold is extremely distorted and loud. It didn't used to be this way and I haven't changed anything, yet it persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can anyone help with this, or has anyone seen this? The mp3s play fine on any computer and haven't changed since they did work. Those wishing to hear for themselves, feel

Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number

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

So I decided, for the formal asterisk rollout, to change over to less commercially-infringing MOH than the prior admin had thrown on the server. (plus: it was blown out and nasty sounding over the phones. Ew.) I changed the files in /var/lib/asterisk/mohmp3 to something else (can't dig up the link, but it was from the voip-info wiki). My musiconhold.conf looks like this: ; ; Music on

I have tried to get MRTG to graph my Asterisk box but have run into a problem. When I run the perl script provided at: http://karlsbakk.net/asterisk/ I get the following error: [root@tsr asterisk]# ./asterisk-mrtg -h myasteriskip.mydomain.com<http://myasteriskip.mydomain.com>-v -1 SIP -2 IAX2 -u 109 -p xxxx Asterisk Call Manager/1.0 Action: Login Username: 109 Secret: xxxx Response:

I was just browsing Digium's web site and noticed they are taking orders for the new IAXy. Has anyone purchased and tested one of these yet?? I have thought about buying one for testing, but want to make sure it isn't going to be a flop like the last one. Robert

I would like to setup a test number, that speaks back my phone number. How can I set this up? bye Ronald

I have asterisk with two zap channels which are analog ports off a T1. They each have a inward DID number If they are used for outgoing they show the T1 main number not the DID's number. Is there any way to send caller ID of the inward DID number not the main number Jeff

I have connected asterisk 1.0.7 with Avaya Definity via E1 with a TE100P Digium Card. Inbound calls are working perfectly and I dont have any problem. But when I try to make an outgoing call with my softphone (xlite) I am getting the following messages. Hungup 'Zap/13-1' Executing Dial("SIP/IZ-bc0a", "Zap/g1/3118") in new stack Called g1/3118 Channel 0/1, span 1 got

Hello All, I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy static. Even with the pots line disconnected, if I do a dial I still get static. This way I know it's not the line, but rather something on the card. I tried alternate pci slots. This card has a power connector, does anyone know what the power requirements are? The unit is in a small case with a 2.4ghz p-4 and

Hi ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference "FM" is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. Thank you in advance for your help Conrad

I am looking into the possibility of buying a Cisco 7960 with a 7914 expansion module. I know a lot of people are using the 7960, but I haven't read much about the 7914 and I was wondering if anybody has used this module with Asterisk? -- Thank you for your time __________________________________ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com

Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Thanks,

Hi All, I am looking for a GSM VoIP gateway for use with Asterisk. I have come across VoiceBlue by 2N but it's price is beyond my reach. Are there any other alternatives out there? I've scanned across the mail achieves for an answer to this without much success, if the question has already been answered kindly point me to the resource. Allan.