similar to: Minor Registration Problem With Polycom Soundpoint IP 500

Hi, Does anyone know the difference between RFC2543 and RFC3261? They are both SIP, but apparently incompatible. We are testing some hardware devices that support RFC3261 and it appears Asterisk is supporting RFC2543 and not completely compatible with RFC3261 (see below). Does anyone know how to configure Asterisk so it can do what is missing? 3. The problem occurs with confirm message

Hi, I am in a predicament and any help/pointers would be appreciated. We are using chanspy to listen in on conversations. We are doing this via a web interface. The web interface lists all the ongoing calls. We click on a call and then my local phone rings allowing me to spy on the session I clicked on. But "most" of the time, when I start listening in, the two parties that are in

Search google with "sip pstn site:www.microsoft.com" You will find out how to configure LCS static routing to SIP Gateway, like Asterisk but you need patch Asterisk to support TCP. http://bugs.digium.com/view.php?id=4903 Step1: configure LCS 2005 to let sip uri: *@pstngw.domain to route to next hop: pstngw ip address Step2: patch your asterisk chan_sip.c to support TCP Step3: configure

hello, I would like to realize a VoIP Gateway, with some extra-features. The aim is to get the phone number of the caller, to make research in our database, and to put him automatically through the good employee. The company is equipped with a VoIP network : software : - PSTNgw - Ohphone - OpenGatekeeper hardware : - Quicknet phonejack - gateway : Voicetroniw OpenLine4 Is it possible to

Hi, I require some urgent advice/help concerning chanspy. We have a web interface that lists all bridged SIP calls (show channels concise) When we click on the call, our local extension rings. When we pick up to spy, parties involved in the call that we are spying on are suddenly unable to hear each other. This happens as soon as we start spying... why is this happening? We are using asterisk

Can multiple FWD accounts be registered? I have the following output in my sip.conf file: register=74928:xxx@fwd.pulver.com/74928 register=75160:xxx@fwd.pulver.com/75160 register=74573:xxx@fwd.pulver.com/74573 [fwd-74928] type=friend secret=xxx username=74928 host=fwd.pulver.com [fwd-75160] type=friend secret=xxx username=75160 host=fwd.pulver.com [fwd-74573] type=friend secret=xxx

Hi, Some calls I make trough my PSTN asterisk gateway just hangup after some minutes. Even if I'm using sip or iax. I have callprogress=no busydetect=no in my zapata.conf. Anyone help? Or tell me what to look at /var/log/asterisk/debug. I didn't find anything wrong. [endpoint]---iax or sip----[asterisk]----E&M----PSTN. As endpoint I had tested another asterisk box (with a FXS),

Hello, I am trying to run festival, it is running but I am getting this when I run tts_ping.agi WARNING[278546]: File app_festival.c, Line 304 (festival_exec): Text passed to festival server : Enter the eye-p address you wish to ping. WARNING[278546]: File app_festival.c, Line 381 (festival_exec): Passing text to festival... WARNING[278546]: File app_festival.c, Line 400 (festival_exec):

Has anyone ever seen these errors generated by a cisco 7960? none of our other brand phones seem to generate these erros: Jan 27 21:54:07 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie s exceeded on call 000ded24-d7000024-5d2ca17a-29c81cf4@65.204.176.54 for seqno 1 01 (Response) Jan 27 21:54:08 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie s exceeded on call

All, Here's a cool one.. I was attempting to call a retarded conferencing service, and was having problems with it picking up my DTMF.. after trying all the settings my Sipura SPA2000 offers, I found inband actually works.. unfortunately, I can't get anything else to pick up my inband DTMF (including asterisk's builtin voicemail! It just times out and says I never entered a login!).

Hey, all. I'm in the middle of a rollout, and just learned that the SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued. The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130 more/handset. AND it doesn't look as nice. Ouch. Does anyone have any recommendations -- Polycom or otherwise -- for a good-quality, mid-range, two-line SIP phone (with good

Hello! I am selling hardware I have been using in various Asterisk test installations. If you are interested in buying all or any of the following items please reply off list. I got a few * X100P clones * Sirrix PCI4S0 ISDN BRI cards (+ 1 Sirrix S0 voltage/power supply) * 1 Polycom Soundpoint IP 500 (+ Plantronics headset) * Conrad HFC-S cards to give away. Some of the pieces are unused,

Does anyone have the Admin password for the phone in order to change configuration Roman

I've looked all over.. I can only find the polycom soundpoint ip 300 w/ sip @ atacom but they're all out of stock. Any input would be appreciated.. Thanks.

I've taken a look at: http://www.voip-info.org/tiki-index.php?page=Getting+MWI+on+Polycom+Phon es+to+work+with+Asterisk I've followed those directions but when I press my message key it doesn't dial the number specified... It actually dials 614p which is the user I register with. My MWI light works, and worked before following those directions, but... I can't make it auto

now that I have finally figured out what I was doing wrong with my polycom phone and got it to read the configuration file Im changing some gains. I successfully changed the gain for the ringer... It was too low for me. Does anyone know which gain would be for the call waiting and which tone would be for the "hands free" mode? Thanks

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

The piece of plastic is built in to the IP600. Use a screwdriver or something similar and push the little piece of plastic out from the top inside the cradle. It will pop out, and you can turn it over and reinsert it upside down to hold the receiver in place. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On

What headsets have people found work well with the Polycom Soundpoint 500 & 600? If any of the headsets that work with the Polycom's also work the Cisco, that would be an added bonus, but is by no means a requirement. I am not looking for a headset that requires an external amplifier, but rather a headset that can make use of the headset jack on the phone itself.