similar to: Upgrading to Asterisk 13 - Strange Music On Hold Issue - Banging my head on this one

Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34

sorry... typo.... the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am

Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next

Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound?

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.

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT.

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

http://bugzilla.mindrot.org/show_bug.cgi?id=764 ------- Additional Comments From kees+bugzilla-mindrot at outflux.net 2004-01-07 12:49 ------- Created an attachment (id=523) --> (http://bugzilla.mindrot.org/attachment.cgi?id=523&action=view) Patch to add configurable version information This patch provides the following new fields in sshd_config: ProtoVersionMajor ProtoVersionMinor

I'm setting up a new Asterisk system on a Dell server and I'm getting "fuzzy" voice between the Polycom IP SoundStation 550 and the Asterisk server. I've checked all of my codec settings and both the Asterisk and the Polycom agree on u-Law encoding. I'm using the latest release of the Asterisk code (1.4.17) and other software. If I call between phones (i.e. two

I would like to know if any body is using the Polycom Soundstation IP 4000 SIP conference phone with Asterisk. I am thinking of purchasing one. Kurt

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone successfully using the Polycom SoundStation VTX 1000 with Asterisk? I can't see any mention of it on the wiki page: http://www.voip-info.org/wiki-Polycom+Phones Thanks, Alvin

Announcing a new Windows-based IAX/IAX2 client. Please download it and give it a try. Let me know about any bugs, and any missing features. I have yet to come up with a catchy name for it, so at this point it calls itself IAX Phone. (Suggestions? Non-derogatory suggestions, preferably). Download: http://www.sokol-associates.com/Downloads/IaxPhone.zip Reference & Support Page:

Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Thanks -------------- next part -------------- An HTML attachment was

All; I am running Asterisk 11.6-cert13 and am having a problem when parking a caller. A call comes in and I put them on hold with *1 (defined in features.conf) without a problem. I can then dial the parked call extension number, say 701, and retrieve the call. The problem I'm having is if I try to park the call again, nothing happens. I hear a beep when I hit *1, but thats it. Has

So when someone's brute forcing your server is there a way to identify the originating IPs without using a tcpdump? When I get a failed auth on the console it shows 'account at asteriskserver' then tag=as25ca5023 (or some random string, though it's a bit odd as alwaysauthreject = yes is on in sip.conf). Anyway, the logs don't show anything more useful either. Is there

I have a polycom 501 phone that is 1 hour off now. Before last sunday (time change) the time was fine. <?xml version="1.0" standalone="yes"?> <PHONE_CONFIG> <OVERRIDES _.0x20._log.level.change.sip="0" tcpIpApp.sntp.daylightSavings.stop.date="4" tcpIpApp.sntp.daylightSavings.stop.month="11"

Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and

I have InAccess Networks' oh323 installed and partially working. I can call the h.323 phone from asterisk using Dial(oh323/${IP_ADDRESS}). How do I dial from the phone to an asterisk extension? It does not appear to me that the phone actually registers (or attempts to register) with asterisk. I'm using Asterisk Stable and the phone in question is a polycom Soundstation IP 3000 or

We have extra equipment that was over-ordered or unused. All of the equipment is brand new. The equipment has been highly discounted to move quickly - the last set of equipment sold in 48 hours. If this equipment is of interest to you, call or e-mail quickly. Buy on VOXILLA and SAVE $300 each (Cisco routers & switches): http://store.voxilla.com/customer/home.php?cat=259 For Sale (all new):