similar to: Polycom phone locks up, send sip busy messages

I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has

I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070328/88e22671/attachment.htm

I have compiled zaptel on Mandrake following everything I have always done on Fedora. It is 2.6 udev so... I had to modify the 01-devfs.rules Make linux26 Make Make install... Everything appears to compile correctly but it says module not found when doing "modprobe zaptel" Is this a rights issue? Jordan Novak -------------- next part -------------- An HTML attachment was

On 6/4/07, Jordan Novak <jnovak@logisticshealth.com> wrote: > I am having a difficult time with the transition from agentcallback login... > Here are a few of the isssues, I am logging in using chan_ local > ie:local/8000 as the extension I'm not sure if this will solve any of your problems or not, but I've found it's often necessary to use the "/n" on the

We are using an Asterisk box to do conferencing right now. I have had about sixteen active lines in conference and the quality was acceptable. We now have a need for 50 people to conference at one time. Does anyone have enough experience doing this to give me some pointers. Will it even be reasonable to try this? Is the mixing done on the the hardware, I plan on using a quad span t-1 card from

I am looking at some directions on how to install and it is asking me to edit defines.php, it states that the file should be located in the source directory, but I can't seem to find it anywhere on my machine. Anyone been thru this? Jordan Novak Communications Technician -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have three polycom 501 that are all hearing echo. The other party sounds fine but you can hear yourself rather well. The volume does help if lowered but that also makes the other party extremely quiet. Is there any way to control the gain of the mic or stop the microphone from picking up so much from the handset. It only happens while you are on the handset. -------------- next part

I have installed asterisk on numerous servers. Every install was done on Fedora and (White box Linux). I now have zap channels in one of the boxes (T-1). No matter what type of channel I call on (sip or zap) I get some really strange artifacts in the sound, almost like a skip in the playback. It seems to always be in about the same place in the recording. Usually in the beginning of playback. For

Is there an app or softphone for meetings that displays the hosts screen like webex or intercall. Jordan Novak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060321/df90d527/attachment.htm

I agree with these fella's, this is a piss poor way of fixing it. I only know of one call center that used static agents, mostly because they were sold a peice of crap and they had no idea how to use it the other way. I think you will find the majority of call centers are callback centers. This decision has taken Asterisk out of the realm of providing reasonable call center solutions. VIVA

I am using Asterisk to send Voicemail out as Email. I am running into a problem I believe to be caused by the exchange server requiring SMTP authentication. I cannot get the sys admin's to turn it off. Does anyone know enough about sendmail to help me. I am assuming that the default mail client is sendmail. It will also send to other non-SMTP authenticated servers. Your help is much

Hi, I'm trying to configure a Polycom IP Phone SoundPoint 500 to connect it to my Asterisk PBX but with no success. First of all, I downloaded the SoundPoint IP SIP Administration guide I found on internet and then I tried to make a boot server creating an FTP account on my Mandrake 9.1 Linux box but I needed the following files: 000000000000.cfg sip.cfg phone1.cfg ipmid.cfg sip.ld so I

Thanks, I was afraid to try and change gains that I didn't know what they did simply because I don't want to blow a speaker or something.. :) I'll try it today. The only thing I haven't figured out is how to set a default ringer in the configuration file, set my time to EST w/ Daylight Savings and when receiving incoming calls if it's possible to see NAME & NUMBER instead

If that was possible, that would make my life easier as well :) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Graves Sent: Tuesday, August 31, 2004 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom SoundPoint... Gains - Whichis for speakerphone

John, By chance do you know how to set a default ringer? What I have done is the following: <DEFAULT se.rt.1.name="Default" se.rt.1.type="ring" se.rt.1.ringer="7" se.rt.1.callWait="6" se.rt.1.mod="1"/> As you can see, I want 7 to be the default ringer for line 1... For some reason, it doesn't take these changes.

Hi We've setup an Asterisk PBX recently and I encountered the following problem: When [mac address]-registration.cfg file includes the FQDN of the Asterisk PBX for the Polycom SoundPoint 501 phones it will not (even try to) register with the Asterisk PBX unless the DNS (it asks) successfully resolves the name: _sip._udp.[Asterisk FQDN]. Did this happen to anyone else? PS - The

Hello everyone, I have just received 3 brand new Polycom SoundPoint IP 600 from "voipsupply.com" and I have the exact same problem on all of them. When I receive a call, the phone is ringing correctly but when I answer it, it takes exactly 10 seconds before I can hear the caller. I also have SoundPoint 300 and 301 and I don't have that problem with those. I'm using Asterisk

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from

I've got a couple Shoreline IP phones, their Shoreline model number is Shoreline IP 100. I believe this is actually a Polycom SoundPoint IP 300 phone. I believe the phone is using a MGCP stack. I want to use it for testing with Asterisk. 1. I suspect I need to re-image the phone to make it work with *. 2. How can I preserve the current image on the phone? 3. What is preferred image to use