Displaying 12 results from an estimated 12 matches for "jonnyhashem".
2005 Aug 11
2
Sip ports
i have added port=5060 to sip client configuration but
it seems the same problem and in the same errors:
Aug 11 10:29:18 WARNING[9869]: chan_sip.c:843
retrans_pkt: Maximum retries exceeded on call
04b3ccd87e45e719588c54a4017e3b99@172.16.180.21 for
seqno 102 (Non-critical Response)
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2005 Sep 23
2
asterisk invitation problem
when i send calls from an asterisk box to a voip
provider the call fails and give me these messages:
*CLI> Sep 23 21:32:32 WARNING[14595]: chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to '"asterisk"
<sip:asterisk@195.112.214.99:5070>;tag=as19e688a1'
-- SIP/call-0f60 is circuit-busy
== Everyone is busy/congested at this
2005 Jul 19
12
Best VoIP provider
It does not look like Nufone is still in business, judging from the
content on their site, which is very little. There is not even a
configuration document to download, to connect to their network.
The rates file is only for US/Canada calling. No international
rates on this rates.csv file.
I have signed up with a $5.00 account with them way back in November
2004. After signup, I havent received
2007 Jul 02
5
softphone with g729 codec
Hi:
Iam looking for a sip softphone that supports g729 codec
Any one have an idea ?
Reagrds;
jonnyhashem
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2005 Jul 15
1
problems with tdm11P
when i display in extensions.conf this :
exten => s,1,Answer
exten => s,n,DeadAGI(astcc.agi,${CALLERIDNUM})
exten => s,n,hangup
when a call comes the zap doesnot rerad the callerid
an give me this:
Jul 15 10:45:24 WARNING[6693]: chan_zap.c:5739
ss_thread: CallerID returned with error on channel
'Zap/4-1'
excuse me for my english iam italian ,thanks in
advance.
2005 Aug 30
0
astcc hangup problem
does anyone had an experience with not hanging up the
call in astcc, thats my problem sometimes call does
not hang up automatically and even when pressing star
the call continue without stopping.
Regards;
jonny
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2005 Oct 02
0
iax invitation problem
i have opened an account with callshopcompany,and
when ive tried to send calls by the sip i had a
message show an asterisk invitation problem i had
these sip configuration:
sip.conf
[callshop]
type=peer
host=sip.callshopcompany.com
username=XXXXX
secret=XXXXXXX
Then i tried to add these lines and it worked :
sip.conf
[callshop]
2006 Feb 16
2
show calls
HI:
what is command on console to know how many calls are
sending in the same time?
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2005 Oct 16
1
iax invtation problem
i had a sip invitation problem with my voip provider
and here the message that was shown :
Oct 16 20:23:19 WARNING[21901]: chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to '"asterisk"
<sip:XXXXX@195.112.214.99>;tag=as7b43dfbd'
-- SIP/callshop-3fcc is circuit-busy
== Everyone is busy/congested at this time
-- Got SIP
2006 Mar 30
2
TDM04B sound volume
HI:
Is there any way to raise up sound volume on fxo on
TDM04B without changing tx-gain and rx-gain ?
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2005 Aug 09
0
channel_pvt.h not found
after editing the Make file i tried to compile
asterisk-oh323-0.6.5 but many errors displayed like
this:
[root@localhost asterisk-oh323-0.6.5]# make
for x in wrapper asterisk-driver; do make -C $x build
|| exit 1 ; done
make[1]: Entering directory
`/home/ali/asterisk-oh323-0.6.5/wrapper'
./check_ver /home/ali/pwlib pwlib
./check_ver /home/ali/openh323 openh323
ar rc liboh323wrap_s.a
2005 Jul 26
1
problems with compiling asterisk-oh323-0.6.5
after editing the Makefile according to my directories
ive compiled asterisk-oh323-0.6.5 but i got these
errors can any body help me with this :
[root@terra@net asterisk-oh323-0.6.5]# make
for x in wrapper asterisk-driver; do make -C $x build
|| exit 1 ; done
make[1]: Entering directory
`/home/wassim/asterisk-oh323-0.6.5/wrapper'
./check_ver /home/wassim/pwlib pwlib
./check_ver