Displaying 20 results from an estimated 2000 matches similar to: "Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)"
2005 Jan 14
1
handle_request registration failed?, Polycom IP500
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
2006 Dec 11
1
Asterisk Sends 180-RINGING to UA even with progressinband=yes
I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ?
Doug.
2006 Dec 11
2
Asterisk Sends 180-RINGING to UA even withprogressinband=yes
Andrew,
I don't think it's a Polycom issue. We took Asterisk out of the picture and had our Polycom phones communicate directly with an Audiocodes PSTN gateway. Unlike Asterisk, the audiocodes do not send 180 Ringing before sending 183 Session Progress, and the polycom's play the correct tones in this case.
We WANT Asterisk to send progress tones in band. In our case it IS needed.
2016 May 03
3
Migrating asterisk 11 to 13: some callers get no ringback tone any more
Hello!
I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2.
As customer of German Telekom, I have three numbers and therefore three
trunks - each number is bound to one trunk.
After migration, some callers complained about missing ringback tone:
they didn't hear any ring tone and where surprised that they suddenly
got me anyway :-). The connection afterwards was as expected.
Deeper
2005 Aug 19
2
Asterisk and Vonage - Can't call out but can receive calls
Hi,
I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me to place and receive
calls. I have successfully configured Asterisk to route inbound calls and send
them to the correct extension, but I can't get outbound calls to work. I have
Asterisk successfully registering with Vonage, but when an INVITE is sent out, I
get a "404 Not Found" back from Vonage
2016 May 03
2
Migrating asterisk 11 to 13: some callers get no ringback tone any more
Whoops, email client auto-filled dev previously. Let's try this again.
Michael Maier wrote:
<snip>
> Ok - but this doesn't seem to answer my main question:
>
> Why must
>
> progressinband=never
>
> be applied especially if asterisk uses it by default? The big difference
> between w/ and w/o it is:
The default in 13 is "no" which still
2019 Jun 14
2
Early Media Issue
Hi all
I've got an issue where when I call a number that just plays early media
back to me.
Instead of hearing the full sequence of tones I hear a short ringing then
part of the sequence. What seems odd is that I can see
the telephone-event/8000 being passed up the chain but when it gets to
Asterisk, it is never sent back to the phone. Instead I just see the usual
RTP flows.
I've been
2014 May 07
1
early media (video)
Hi All,
I've been looking for information on how to use asterisk and early media to
allow for a video-preview of the caller at the callee's phone for days...
but I haven't been too successful :(
I found that there seems to be a company "2N Helios IP" which claims
(youtube-video) that "their" SIP server is able to provide early video
(using a Grandstream 3157v2
2009 Jun 13
2
Polycom registration errors
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
2006 Mar 22
5
Double Call Progress tones
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
This is slowly driving me nuts!
I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk
1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls
I get a double ring tone (UK style + US style). I also have a DECT phone
on a Sipura SPA-3000 configured with UK tones. This gives me a double
ring of UK + UK, so this
2005 Mar 26
1
DTMF tones not working
I have Polycom ip-300 phones that worked yesterday but dont seem to work
today (at least dtmf signalling once connected to the asterisk box)
The current configuration is:
[general]
port = 5060
bindaddr = 0.0.0.0
context = test
srvlookup = yes
dtmf = inband
allow = all
dtmfmode=inband
progressinband=no
disallow=all
allow=ulaw
pedantic=no
[202]
type=user
secret=xxxx
context=test
mailbox=202
2004 Jul 18
4
Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All
Total noob on the list so all help appreciated....
I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows).
I've plugged in two Cisco 7960 phones....
The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......
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:
2007 Jun 22
1
Does Early Media have to be Ulaw?
I have this in sip.conf:
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
progressinband=yes
[19256002182]
type=friend
username=19256002182
callerid="Test hone 1" <+19256002182>
host=dynamic
canreinvite=no
secret=password
context=test
disallow=all
allow=g729
[level3]
type=peer
host=xxx.yyy.16.99
context=default
2004 Dec 29
1
Polycomm IP500 dropping incoming calls
Hello everyone.
I can place outgoing calls no problem with my IP500 (using teliax as our
provider). Thing is, when a call comes in, 90% of the time when I pick up
the handset it drops the call immediately. I turned on SIP debug, and have
listed my extension config from sip.conf. Any help is greatly
appreciated.... sooo close.... TIA! -Ron
[3004]
type=friend
username=3004
password=XXX
2007 Feb 13
2
Customisable In-band ringing?
All,
Using SIP with progressinband=yes I get Asterisk to generate the ringing
sound for callers. However, I was wondering if it is possible to
configure what is 'played back' to the calling party? i.e. instead of
just 'ring ring' could I potentially play back a song from an MP3, WAV
or GSM file? I'm thinking it would be quite cool to offer a customised
'ring'
2014 Jan 15
1
How to tell Asterisk to to send Ringing signals as into RTP
Hello,
My target system is :
PSTN <---> Sip Provider <---IP/ADSL---> Router with fw/NAT <--- SIP/IP/Eth
--> Asterisk <--- SIP/IP/Eth --> SIP Phones
Asterisk is configured to keep NAT connection with SIP provider open (with
qualifyfreq) so I don't have any problem (yet) with either casual incoming
or outgoing calls.
To work around a possible No Audio when an incoming
2006 Feb 23
3
Polycom IP601 Question
Hey everyone, I haven't seen an issue quite like mine, so I am hoping
anyone who used the Polycom 601's may have an idea.
We are going to be switching our office over to Asterisk. All the phones
are going to be 601's, I am going to set up a boot server, but for now I
am just going to test everything on one phone. My question is I have the
phone registered in Asterisk (phone icon
2005 May 12
0
Asterisk, SIP and NAT: Help needed!
I've been googling and talking with Libretel about my
setup and the fact that incoming calls to my asterisk
box through the Libretel number reach my box (I hear
the greeting being played) but then don't accept DTMF.
Here is a rough diagram of my setup:
Asterisk |
server | NAT <------------ Libretel
| router
|
Note that there are NO SIP
2014 Jul 26
1
Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
Greetings,
I've noticed a problem that might originate from my Asterisk configuration,
could use a hand in sorting it out. Problem is a 488 response from Asterisk
whenever it gets RTP/SAVPF profile in the SDP.
My current setup has Asterisk Kamailio realtime integration, and Kamailio
uses dispatcher to route calls for Asterisk to handle. Now I have only one
Asterisk, on the same machine as