Displaying 20 results from an estimated 2000 matches similar to: "SIP codecs"
2003 Jun 06
1
more about SIP ...
I added the line "allow G723.1" in my sip.conf general config,
and from a bridge connection which gives silence,
I have progressed to the error message below,
and the call gets rejected.
help!!
Dave
ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant
Expressa
723@216.52.153.207 : Go2Call SIP gateway
-- Executing
2003 Jul 08
2
oh323 problem (small one)
I have just compiled & installed the latest oh323, on a fresh asterisk
installation
however using a previously working oh323.conf file.
When I try to dial an outbound oh323 call I get the following error :
-- Going to extension s|1 because of immediate=yes
-- Executing Wait("Zap/1-1", "1") in new stack
-- Accepting call from '21382890' to 's'
2003 May 27
1
please help (reposted) - re. * connecting to a commercial call service
hi,
maybe someone out there already has some experience and can help me.
I have just ordered an E100P card from Digium, I already have a basic
asterisk setup up & running.
My application is the following :
I want to accept incoming calls from the PSTN to Asterisk, and without
asking anything of the client just pass them immediately to a call gateway
in USA, actually we are planning to use
2006 Dec 12
1
AGI problema
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<title></title>
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<body bgcolor="#ffffff" text="#000000">
<font face="Verdana">Hi all. I've written a AGI in C language.
2005 Mar 19
2
Problem Making a SIP call over a long latency network - Call rejected: 407 Proxy Authentication Required
Hello,
We are getting error: Call rejected: 407 Proxy Authentication Required - if
a user is trying to call using * over a long latency network (around 600
ms). There is no problem when the same user is trying to make a call with
low latency network (around 300 ms). I have included the debug and log
messages for Asterisk. This call is done with SJphone, the same problem
exists with ATA;
2005 Aug 13
14
Why NAT problem
hello
i am using asterisk-1.0.9. i have a NAT problem.
without NAT registration is ok. and if user is bhind
NAT it is registring on asterisk. but SJPhone is
showing "not registered". i think asterisk is properly
sending request to UA. any comments............this
sip.conf setting was working previously
-- Registered SIP '5000' at 0.0.0.0 port 5060
expires 120
-- Saved
2004 Sep 22
4
Softphone for PocketPC or iPaq
Is there a soft phone for PocketPC or iPaq? If not, is someone working
on it? I will be more than willing to contribute my mite if needed.
Thanks,
-- sudhir
2006 Nov 02
1
is IAX required for firewall and router?
I'm trying to understand IAX and whether or not it would solve my
difficulties:
'The primary goals for IAX were to minimize bandwidth used in media
transmissions, with particular attention drawn to control and individual
voice calls, and to provide native support for NAT (Network Address
Translation) transparency. Another goal is to be easy to use behind
firewalls.'
2005 Feb 27
3
music on hold trouble
Hi All
I seem to have a small problem with the music on hold button on SJPhone.
I have 2 asterisk installations one from the Rapid distribution and one from the latest CVS.
On the rapid dist when I press the music on hold button on my SJPhone I get music on hold.
When I do the same I get no music on hold just silence.
I create extension like this exten => 1111,1,MusicOnHold(Default),
2007 Jan 17
4
windows mobile 5 softphone for square screen devices
Guys, anybody has seen or is using some kind of softphone on any square
screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they
do work on Wm5 but they are designed for standard screens, anybody using
anything on square ones?
2004 May 25
4
Sip/IAX Clients for Linux
Hi There,
i think all VOIP clients for Linux are unusable!
i got testet:
Linphone + Linphonec all in version 12.2
Kphone
gophone
and other...
the only programm that is usable is gnomemeeting...
does anybody knew some other tools?
Best Regards,
Mark
2004 Dec 09
2
hfc card and isdn error E001B
I'm trying to use an hfc based pci card with asterisk but every call fails
falling in the congestion extension.
exten => _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr)
exten => _0.,2,Congestion
Looking in the syslog i can see:
isdn: HiSax,ch0 cause: E001B
it seems that this is a terrible error when arrives... hard to tell what is
the cause. Also terrible is finding a lot of material
2003 Oct 13
1
chan_h323 - Segmentation fault (core dumped)
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used for test es SjPhone
(http://www.sjlabs.com/).
This is the data for one core dump:
(gdb) bt
#0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790
#1 0x41f8879c in create_connection
2003 Jul 10
2
OH323 + G729 + Go2Call
hi ..
i've just installed and licensed an instance of the G729 codec.
I am trying to connect through asterisk to Go2Call server ..
According to their info it involves dialling extension 729 on
voip01.go2call.com, to get the IVR.
my extensions.conf shows :
exten => s,2,Dial(OH323/h323:729@216.52.153.206)
which I think is correct, I have G729 enabled in the OH323.conf
file and it seems to
2003 May 13
1
beginner's question!
hi there,
I have just downloaded and installed asterisk a couple of days ago, it compiled correctly and starts up and runs, on a Redhat 9 system freshly installed for testing. I don't have any extra hardware installed so far, was attempting to just try out connectivity. I am having some probs with the configuration, maybe someone out there can give me some tips :
firstly on modifying the
2004 Jun 17
3
SJphone regestration problem - Help!
I am having a problem with SJphone registration, having read the list
and wathced it for a while for similar problems. I just can't seem to
figure out the problem.
I tryed to follow a tutorial from
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone,
but in SJphone (SIP tab), I can't find the following setting.
Use local outbound proxy - checked.
Proxy IP Address:
2003 May 26
1
Quetsion about DISA...
Hi all,
i use the DISA app for giving the user a trunk after a authentication
through PGSQL as follows
.... auth via PGSQL
exten => s,1,DISA,no-password|test
I think the user is now in context "test" and he could dial any number if
the extension-conf in "test" is for example
exten s,1,Dial,OH323/<myip>
But if the user dial one digit the call build up
2003 Jul 29
3
stupid questions ..
just three "stupid" questions I need to ask ..
1. what's the sequence to press on a SIP phone to transfer a call to another
extension.
2. what's the same thing if you want to hold an incoming call, speak to the
other extension, then pass the call?
3. what's the extensions.conf syntax to dial two SIP extensions at once?
many thanks
Dave
2003 Sep 13
2
SJphone DTMF?
Hi. I have sjphone installed on windows and working
except for dtmf. I read the docs for sjphone and it
uses inband dtmf. I configired dtmfmode=inband but it
still does not recognize it. Someone on the lists
said that inband only works using alaw or ulaw but i
tried only allowing that too but still no go. Hmm..
any other ideas? I can't get any other client to work
on windows :-/
I
2005 Oct 08
1
need help-can't not register to asterisk from softphone
Dear all expert,
(i posted this question one time, but i couldn't reach the answer
-so allow me to post here)
1)I download asterisk realse version 1.2 beta1.
After that i compile it successfully and run it with:
asterisk -vvvc
2)I follow the instruction in
http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html
in sip.conf:
i add two account:
[ivan]
type=friend
username=ivan