search for: voipprot

...uot;" LOG_FILE_DIRECTORY="/log/" /> --------- sip.cfg --------- <?xml version="1.0" standalone="yes"?> <!-- SIP Application Configuration File --> <!-- $Revision: 1.63 $ $Date: 2004/11/08 18:52:16 $ --> <sip> <voIpProt> <local voIpProt.local.port=""/> <server voIpProt.server.1.address="10.0.20.0" voIpProt.server.1.port="5060" voIpProt.server.1.transport="DNSnaptr" voIpProt.server.1.expires=&quo...

...tty 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 does (at least conceptually) but ss the "x" variable still referring to a server (1 or 2)? And the "y" var, what is it referring to? An event? Which one? "Determines which events specified with the Event he...

....userId="23" reg.3.auth.password="23" reg.3.server.1.address="10.20.0.1" /> </phone1> And sip.cfg: <!-- IP Application Configuration File --> <!-- $Revision: 1.57.2.1 $ $Date: 2004/07/27 00:23:34 $ --> <sip> <voIpProt> <local voIpProt.local.port="5060"/> <server voIpProt.server.1.address="10.20.0.1" voIpProt.server.1.port="5060" voIp Prot.server.1.transport="UDPonly" voIpProt.server.1.expires="3600" voIpProt.serv er.1.register="1" voIpPr...

...hese topics it would be greatly appreciated! Many thanks, Mike MY CURRENT SIP.CFG: ------------------------------------------------------------------------------- <?xml version="1.0" standalone="yes"?> <!-- SIP Application Configuration File --> <sip> <voIpProt> <local voIpProt.local.port="5060"/> <server voIpProt.server.1.address="10.215.100.1" voIpProt.server.1.port="" voIpProt.server.1.transport="UDPonly" voIpProt.server.1.expires="3600" voIpProt.server.1.register="1" v...

Hi, I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header. Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it's worth my time

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

Hello all, I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that blindpreferred=1 (all transfers default as blind transfers). If a customer calls in & we answer & transfer, everything works fine. But if we call out to a customer & then transfer to another internal extension, that

..." reg.1.acd-login-logout="0" reg.1.acd-agent-available="0" reg.1.proxyRequire="" reg.1.ringType="2" reg.1.lineKeys="3" reg.1.callsPerLineKey="1"/> Excerpt from sip.cfg: <server voIpProt.server.1.address="192.168.35.1" voIpProt.server.1.port="5060" voIpProt.server.1.transport="DNSnaptr" voIpProt.server.1.expires="" voIpProt.server.1.expires.overlap="" voIpProt.server.1.register="1&qu...

...hours looking at many, many web pages that all say the same thing. I am using Asterisk 1.6.2.18 and Polycom phones, both older (501 with "latest" legacy 3.1.7 firmware) and newer (335 and 650 with latest 3.3.1f). I have changed the correct values in sip.cfg like this: <alertInfo voIpProt.SIP.alertInfo.1.value="Ring Answer" voIpProt.SIP.alertInfo.1.class="4" /> and <RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod=&q...

...doesn't seem to have a lot of information about this kind of thing. In your dialplan, you'll need to set the _ALERT_INFO variable to whatever you'd like your ring to be. Your choices are in sip.cfg in two places: Right at the beginning, you'll see a line like this: <alertInfo voIpProt.SIP.alertInfo.1.value="" voIpProt.SIP.alertInfo.1.class=""/> And then later on, you'll see some lines that look something like this: <ringType se.rt.enabled="1" se.rt.modification.enabled="1"> <DEFAULT se.rt.1.name="Defa...

In ipmid.cfg I have: <G3INTERCOM se.rt.10.name="G3INTERCOM" se.rt.4.type="ring-answer" se.rt.4.timeout="1000" se.rt.10.ringer="7"/> In sip.cfg I have: <alertInfo voIpProt.SIP.alertInfo.1.value="G3INTERCOM" voIpProt.SIP.alertInfo.1.class="10"/> I set up a test extension: exten => 8614,1,SetVar(ALERT_INFO="G3INTERCOM") exten => 8614,2,Dial(SIP/614p) Ringer isn't changed.. In addition I have tried using the <ringType&gt...

...2.label="1003" reg.2.type="private" reg.2.auth.userId="1003" reg.2.auth.password="" reg.2.server.dnsLookupOption="0" reg.2.server.1.address="192.168.2.2" reg.2.server.1.port="" The local overrides file for the phone contain: voIpProt.server.2.expires.lineSeize="" voIpProt.server.2.retryMaxCount="" voIpProt.server.2.retryTimeOut="" voIpProt.server.2.register="" voIpProt.server.2.expires="3600" voIpProt.server.2.transport="DNSnaptr" voIpProt.server.2.port="5060&quot...

Hi, My phones stopped auto-answering when being paged, since I moved on to Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk 1.6.2.16. I looked at the wiki but nothing I try there works, even if I cut and paste the same setup. Any one has any idea of what I should change from my old 3.2.3 setup? My older phone (501) still using 3.1.6 still auto-answer

Hi, I'm having trouble setting the ringtone on my Polycom 501. The relevant entry in extensions.conf is: exten => 801,hint,SIP/creative1 exten => 801,1,SetVar(ALERT_INFO="Test") exten => 801,2,Dial(SIP/creative1,20,Ttr) In the sip.cfg: <alertInfo voIpProt.SIP.alertInfo.1.value="Test" voIpProt.SIP.alertInfo.1.class="13"/> and <TEST se.rt.13.name="Test" se.rt.13.type="ring" se.rt.13.ringer="11" se.rt.13.callWait="7" se.rt.13.mod="1"/> It does seem to be sending the proper...

...#39;m trying to make an IP-501 auto answer a call. exten => 301,1,SetVar(_ALERT_INFO="Ring_Ans") exten => 301,2,SetVar(ALERT_INFO="Ring_Ans") # Tried both combinations exten => 301,3,Dial(SIP/5001,15) exten => 301,4,Hangup Sip.cfg for Polycom phone <alertInfo voIpProt.SIP.alertInfo.2.value="Ring_Ans" voIpProt.SIP.alertInfo.1.class="4"/> Ipmid.cfg <RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="1000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod=...

I am looking for clues on how to configure distinctive ring for a PolyCom SoundPoint 300. Does ALERT_INFO apply? If so, how? Thanks, David Koski david.nospham@kosmosisland.com

(new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone?

I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call "22" and the phone rang it did not auto answer. Did I miss something? exten => 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten => 22,n,SipAddHeader(Alert-Info: Ring Answer) exten => 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten =>

...list but to no avail. I have configured Asterisk to send the callerID of extension phones as "firstname lastname" and that seems to work well and extensions show calls originating on other extensions in this format. I set the following in sip.cfg for one of the phones: <alertInfo voIpProt.SIP.alertInfo.1.value="John Smith" voIpProt.SIP.alertInfo.1.class="4"/> So when John Smith calls that phone it should ring-answer as per Polycoms documentation but it doesnt it jsut rings as per usual then goes to voicemail can anyone point me in the right direction? I wa...

...="207" reg.1.address="207" reg.1.label="207" reg.1.type="private" reg.1.lcs="" reg.1.thirdPartyName="" reg.1.auth.userId="ip4000_1" reg.1.auth.password="password" == Polycom company-wide config on TFTP server == server voIpProt.server.1.address="x.x.x.55"/> <SIP> <outboundProxy voIpProt.SIP.outboundProxy.address="x.x.x.55"/> </SIP> I've tried using x.x.x.55 as both the "proxy" value only, the "server" value only, and (in the given example) both. I also add...