similar to: RE polycom fails registration

Hello list, I was wondering if any of you guys have had any luck with polycom in remote offices, I'm facing a weird issue, polycom phones work fine in the main office, in remote office it says, Registration from '<sip:202@10.0.1.190>' failed for '70.59.21.112' - Wrong password the odd thing is Linksys phone works without any issue!! polycom wont register but its able to

Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffrences? At this time i'm using

hi there I'm setting up asterisk@home and I'm using Polycom IP 500 phones. When I call a number that has a digital receptionist (i.e. "dial 1 or such and such, dial 2 for this and that...") the Polycom doesn't seem to send the extra digits. When I try it with X-Lite things appear to work fine, so I think the problem is with the Polycom configuration. Here's some

Hi all, My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. I've started studying the docs and I'm having trouble understanding the following xml attribute: voIpProt.SIP.requestValidation.x.request.y.event I understand what it

I'm planning to upgrade my asterisk 1.4.4 to 1.4.6. usually for asterisk upgrade i delete modules directory and include, then compile the new version. Since i have couple of G729 Licenses on this server installed, would i need to call Digium to reactivate these Licenses? Is there any better and faster way of upgrade asterisk? Possibly without losing G729 License? Thanks!

> Is your Asterisk server listening on port 5061? If not, just change > the > entry to 5060. Also, I'm not sure how your sip.conf is set up for asterisk, but if you've set it up like: [203] type=friend username=blah secret=blah etc... Your Polycom config file will generally look like this. <PHONE_CONFIG> <OVERRIDES

Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

We recently took a few Polycom Soundpoint IP 500 to test out in Asterisk environment. So far it has been good. Call Hold, Transfer, DMTF etc. However, I do notice every now and then the Polycom fails to register with Asterisk. Asterisk console outputs the following: Feb 3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response: Unable to determine sequence number from '' Feb 3

hello all, I'm having problems getting my polycom soundpoint ip 500 working, and was wondering if anyone would be willing to share their config files with me (the polycom configs). I have managed to get my boot server up and running, and the phone successfully updated its ROM, and downloaded the config files i have put together for it (the display shows correctly but the line won't

Hi I am just getting started with asterisk and trying out using the sample files with a Polycom IP500 (latest sip.ld etc) My question: phone status says not registered: /system status/server status "Line 1 poly is not registered" console message says: handle_request: Registration from '<sip:poly@10.1.20.3>' failed for '10.1.20.50' I have tried dialing and

I'm having trouble with Polycom 501 phones that asterisk forgets how to reach them. /etc/asterisk/sip.conf: [general] context=default MusicOnHold=default port=5060 bindaddr=0.0.0.0 srvlookup=no;yes language=en dtmfmode=rfc2833 maxexpiry=600 defaultexpiry=120 [502] type=friend username=502 secret=pass host=dynamic mailbox=502@rm callerid= "Operator" <502> context=rm

Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble on registration(? -- maybe it's on acquiring an IP?) has started again. I still have the old sip.cfg, but can't figure out which option it is. Any help? Thanks! -Ken D'Ambrosio -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi, One of my users is in trouble with his polycom phone hooked to an asterisk server. The phone works fine for a few days, and then disappears from the registered sip peers in asterisk. The user is able to place outbound phone calls, but can't receive incoming calls until the network plug is unplugged/plugged. Working line XXYYZZAA24/XXYYZZAA24 10.50.5.186 D A 5060

Hi Everyone, Configuring a Polycom conference bridge IP 5000 to connect to Asterisk. For some reason I don't see any SIP packets coming in to Asterisk at all. I don't want to use XML or ftp etc for now and just use the Web Interface to get it going with basic features. But the Web UI is a bit confusing with SIP and Line tabs. I have put this on the web interface: SIP > Outbound

asterisk-users-bounces at lists.digium.com wrote on 01/23/2015 10:24:24 AM: > Hello, > > I'm having a problem with a few Polycom SoundStation 6000s. > Everything works fine, but they drop registration to asterisk after > about maybe 30 minutes ? the phone does not re-try to register and > if you try to dial out on the phone it says ?URI Dialing is Disabled? > > Has

> We run a variety of 5000, 6000, and 7000 series Soundstations running > Asterisk 11.6.0 and the phones are at 4.0.3.7562. We do not see these > registration issues. Would you be willing to send the configuration from asterisk for this? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may

I have sent a Polycom IP500 phone to an overseas remote user who has a Speedtouch adsl modem/router. The phone is connected to the 4 port router which performs NAT for the network. The phone rboots, find the server, downloads it's config and tried to register. Then it immediately becomes UNREACHABLE. Jul 26 17:44:41 NOTICE[21026]: chan_sip.c:9138 handle_request_register: Registration from

Hello, I'm having a problem with a few Polycom SoundStation 6000s. Everything works fine, but they drop registration to asterisk after about maybe 30 minutes - the phone does not re-try to register and if you try to dial out on the phone it says "URI Dialing is Disabled" Has anyone else had this issue? I'm running asterisk 11.7.0. This message may be private and confidential.

This is a SER/Polycom question, but I hoped we may have some SER guru's here... I have a series of Polycom phones that are tying to register with OpenSER. The phone sends a REGISTER message, and OpenSER replies with Unauthorised (all normal). The phone re-sends the REGISTER with the credentials, and OpenSER sends Ok. Here's where it goes downhill. The polycom's appearance display

I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name