Displaying 20 results from an estimated 6000 matches similar to: "Sip no voice"
2004 Sep 18
9
No sound
Hello,
I have just set up an asterisk box (Debian unstable) and I would like
to test it with a H.323 application (gnomemeeting). When I call the
demo voice menu, I can't hear any sound. asterisk says that the
soundfile is played:
-- Executing BackGround("H323/ip$212.9.189.172:30005/29597", "demo-congrats") in new stack
-- Playing 'demo-congrats' (language
2005 Jan 03
4
Manager API
Hi,
Where can I find a complete * manager api guide, the one one wiki is missing
informations like the monitor function for example,
Thnx
Serge
2005 Feb 09
5
polycom soundpoint ip 300
hello,
I try to set up two lines per ip 300 phone,
registration is ok but i get Failure to authenticate
407 for subscribe.
Anybody could help me to configure Asterisk in order
to set instant message and presence ?
I've tried with Ondo sip server it's ok !
Regards
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2005 Feb 08
2
Polycom screwed up Messages button in 1.4.1?
I think Polycom has added another feature that nobody wants.
With MWI configured, and a phonexxx.cfg that has this:
<msg msg.bypassInstantMessage="1">
<mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact"
msg.mwi.1.callBack="XXX" msg.mwi.2.subscribe=...>
</msg>
Under 1.3.4 and earlier, the phone would immediately
2004 Dec 31
3
IAX users
Hi,
I do not understand the difference between SIP and IAX, is it only two
different protocols or something more special.
The problem I have is that I've created two users
Aix.conf
register => users1:passwd1
register => user2:passwd2
[user1]
type=user
context=default
secret=passwd1
host=dynamic
[user2]
type=user
context=default
secret=passwd2
host=dynamic
extensions.conf
exten
2005 Jul 18
2
Crazy stuff in latest CVS HEAD
Hi -
I've just been testing out the latest CVS HEAD (as of about 10:00a
EDT today). I'm getting some weird errors. Calls from one sip phone
to another have OK audio in one direction and highly scrambled audio
in the other direction. The console shows this error repeated ad
nauseum during each call:
Jul 18 16:08:03 ERROR[22941]: utils.c:509 tvfix: warning negative
timestamp
2004 Dec 22
2
MWI not working on Polycom Phones
Hi All -
I'm running version SIP version 1.3.4 on various IP300, IP500, and
IP600 Polycom phones. I'm having a tough time with MWI. I thought I
remembered somebody on the list saying that they had it working, but I
can't find it in the archives now. I have all the phones configured
for MWI as specified in the WIKI:
ipdmid.cfg:
up.oneTouchVoiceMail="1"
2004 Sep 16
2
Uniden UIP-200 Multiple line appearances
Hi -
I'm wondering if any has experience with the Uniden UIP-200 phones.
The product info says that the 8 led buttons at the top are all
programmable. Can they be programmed as separate line appearances (ala
Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the
phone capable of multiple SIP registrations?
Also, the post about these phones at voip-info.org mentions some
2006 Mar 16
1
Re: transfers/parked calls + polycom 501
Hi -
> I am not sure what I did but blind transfers do not work. The Polycom does
> not allow me to dial the extension of the person I want to transfer to after
> I hit:
>
> transfer -> blind
I would strongly suggest getting the latest firmware, and using the sample
configuration files with that firmware to set up your phone. This SHOULD
work. If it still does not work
2004 Sep 17
3
how to get caller ID
i cannot see caller ID of the call originated from outside zap channel.
i hv configured both zapata.conf and extensions.conf.
i m right now in india
i think asterisk only supports Bellcore enable caller ID.
so is it the same bug of BT caller ID problem in UK?
or it is the bug of my asterisk configuration?
i hv enabled callerID from my TELCO.
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2004 Dec 15
2
IP Conference Units?
Hi -
We have a couple of large spaces that we'd like to cover with dedicated conference units like the Polycom Soundstation IP3000. We're concerned about adequately covering the spaces, though, one of which is very long and narrow. I wanted to get external add-on microphones for the IP3000, but I've found that unlike some of their other conference products, it does not have this
2004 Nov 30
2
Dual NAT for SIP
Hi,
My installation at home use two NAT translations before it reaches the linux box where Asterisk is running on.
I use DSL with a Wireless router which fwd all packets to an Windows 2003 box an this windows box it NATing the UDP and RTC packets to my linux box.
If I try to connect to it from outside I get this error :
Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum
2006 Feb 23
9
auto provision of IP501 polycom
Has anyone been able to get the IP501 to discover the FTP server IP
address (via dhcp or dns) and download 100% of the config from a
provisioning server?
We are still having to touch each unit to enter the ftp server address
and password, as well as set many of the options that will not take from
the config file.
Have a sample config file you are willing to share?
What is required in
2006 Mar 16
1
Re: transfers/parked calls + polycom 501
This is a dialpaln issue. I solved the same problem recently.
For 4 digit extensions you need to append the dialplan statement in the
sip.cfg configuration file as follows
<digitmap
dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2
-9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/>
Michael
> I am not sure what I did but blind transfers do not
2004 Dec 03
8
Why, why, why???
Help.
Why is it that I can call out from my GSBudgetone SIP phone but the
audio is "one-way'?
Why is it that when I call my asterisk phone number, I get a fast busy?
2005 Sep 14
3
Asterisk 1.0.9 long term stability
I've been evaluating asterisk for quite some time now and am attempting to
create services on it. The system is simple right now. asterisk seems to
look up atleast every week if not more. I am running asterisk 1.0.9 and
would like to find similiar experiences of long term stability.
I attempted to debug it, but my asterisk isn't compiled with all the
possible debugging flags, which
2005 Mar 17
6
Polycom vs. Cisco IP Phones
Hi all,
I am working on building a new VoIP PBX. Looking at the current market
for phones it seems my best "enterprise" options are the Cisco and
Polycom phones. I have some experiance with the Cisco 7940G, but the
process of flashing the phone with the SIP firmware left a bad taste in
my mouth (not to mention the added expense for the phone).
What is the general consensis about
2006 Jan 27
6
Lockups since upgrade 1.2.3 - anyone else? Any ideas?
Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
hours or so.
Since upgrading to 1.2.3, though, the whole system has locked up twice. Once
on Thursday, and then about a half hour ago. The server would reply to a
ping, but no ssh login, no local console login - just locked up. This ain't
good for
2007 Jul 02
5
softphone with g729 codec
Hi:
Iam looking for a sip softphone that supports g729 codec
Any one have an idea ?
Reagrds;
jonnyhashem
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2004 Aug 12
1
AgentLogin issue
Hi
i have an issue getting agentLogin working
/etc/asterisk/queues.conf
member => Agent/1001
member => Agent/1002
extension.conf
exten => 110,1,Wait,1
exten => 110,2,AgentLogin()
now, i call 110 by a firefly client, trying to login in as 1001 agent:
Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060>
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