Displaying 20 results from an estimated 500 matches similar to: "Blind transfers being cancelled by asterisk & hanging up on remote caller"
2007 Jun 09
3
Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event
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
2005 Sep 27
2
Polycom IP 500 - problem dialing extra numbers
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
2006 Jun 21
4
Polycom 601 problems with multiple registrations
I'm stumped on this one and any help would be greatly appreciated.
I'm just trying to get my Polycom 601 to have multiple extensions on it.
For example, on line 1 I want extension 21, on line 2 I want extension
22, and on line 3 I want extension 23. Ideally I'd actually have each
extension appear on 2 lines and therefore filling up all 6. I should be
able to do that with the
2012 Feb 10
3
Polycom firmware 4.0.1 and paging
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
2007 Jun 07
3
Polycom phone registration problem
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
2011 Aug 08
2
Polycom and auto answer
Hi,
I've been meaning to fix my non-working paging feature here for a while, and
I've just spent the last 5 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
2004 Sep 02
5
Polycom SIP INFO & Changing Ringers
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 =>
2005 Aug 02
9
Polycom phones w/ two lines on different servers
Hi all -
This isn't really directly Asterisk related, but has anyone successfully
set up a Polycom phone to register two lines on two different Asterisk
boxes? I can get the first line to register, but the second one does not.
I can still place calls from that second line, which indicates to me the
server, user, and secret are correct. I'm running the newest 2.6 series
firmware with the
2011 Feb 24
2
Paging with Polycom 3.3.x
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
2006 Feb 09
1
Re: Polycom IP501 with Asterisk - distinctive
Hi Andrew -
> I have a need to be able to identify incoming calls based on some factor
> (could be time of day, caller ID, dialed number, it doesn't matter.) --
> Assuming Asterisk can differentiate between the calls I want, how do I inform
> the IP501? There are "only" three line appearances -- I can't simply just
> ring a different appearance since there
2006 Jan 25
4
Setting ringtone on Polycoms
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
2005 Aug 04
1
PolyCom SoundPoint 300 and distinctive ring
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
2012 Aug 13
1
Websockets on Asterisk 11 and SipML5
Hello,
I'm trying to register a user using sipml5 on Asterisk 11. I followed the
instructions here:
http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets
I added transport=ws to my sip.conf file:
[3002]
username=3002
secret=XXXXXXXXX
host=dynamic
type=friend
context=test
disallow=all
allow=g729
;allow=all ; Allow codecs in order of preference
allow=ilbc
2005 Jul 14
5
Polycom Auto-Answer problems
CVS Head from 07/07/2005
I'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"
2011 Apr 07
3
No ringback even though progressinband=yes is set
Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config.
I have set this on the current system & restarted asterisk, but to no avail.
I am using:
AsteriskNOW distro
Asterisk build is 1.6 from AsteriskNOW repository:
2011 Mar 23
4
What is the most stable version of asterisk?
1.2? 1.4? 1.6? 1.8?
Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCSA, Security+, A+
Linux+, Network+, Server+
.
www.impalanetworks.com
P: (505) 327-7300
F: (505) 327-7545
2003 Dec 30
3
SIP phone as intercom
(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?
2016 Oct 15
2
Registered successfully, but after a minute or so no SIP messages anymore
hi,
let me explain in detail, what i have configured and what is happening now:
1st router w724v (Deutsche Telekom AG):
- port forwarding, everything to destination port 51000-55999 to
device with ip 192.168.2.50 (interface of 2nd router)
2nd router Bintec RS353j):
- configured NAT, everything to port 51000-55999 to device
192.168.3.99 (same ports)
other direction is totally open.
I
2008 Apr 14
2
polycom auto answer
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 =>
2013 May 02
1
Building Asterisk 11.4.0-rc1 with PJSIP 2.1
Hello,
I'm working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead of
2.0 due to a crashing issue resulting from ICE.
https://issues.asterisk.org/jira/browse/ASTERISK-21696
Currently, I'm systematically going through each Makefile in every
directory in pjproject and changing the paths that exist in the pjproject
2.0 included with Asterisk, so that I can successfully build