[asterisk-users] Trouble outgoing VOIP Provider Calls
I have a weird problem
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext - Panasonic Extensions No Problems
Panasonic Ext - SIP Extensions No Problems
SIP Ext - VOIP Provider No Problems
Panasonic Ext - VOIP Provider Errors
-- Working SIP - VOIP
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/610-097aee60,
SIP/acevoip/03) in new stack
-- Called acevoip/03
-- SIP/acevoip-097b52c0 is making progress passing it to
SIP/610-097aee60
-- SIP/acevoip-097b52c0 is making progress passing it to
SIP/610-097aee60
== Spawn extension (from-sip, 903, 1) exited non-zero on
'SIP/610-097aee60'
-- Not Working Pana - VOIP
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/31-1,
SIP/acevoip/03) in new stack
-- Called acevoip/03
[Jan 29 11:00:36] WARNING[20642]: chan_sip.c:11731
handle_response_invite: Received response: Forbidden from 'Unknown
sip:[EMAIL PROTECTED];tag=as3a292a14'
-- SIP/acevoip-097b1358 is circuit-busy
--
Both numbers dialled were exactly the same (9 is the leading number on
all calls in the system and is stripped before dialing), I just replaced
the numbers with .
Tested from several different sip phones and Pana handsets, and it is
only with outgoing calls to VOIP, incoming that go to a Pana extensions
work fine.
--- Extensions.conf
[dialstring]
exten = t,1,Dial(Zap/g1/100,60,tn)
exten = i,1,Dial(Zap/g1/100,60,tn)
[from-e100p]
include = dial-sip
include = out-voip
[dial-e100p]
exten = _1XX,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_1XX,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(
num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN})
exten = _1XX,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _1XX,4,Dial(Zap/g1/${EXTEN},90,r)
exten = _91XX,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_91XX,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID
(num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = _91XX,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _91XX,4,Dial(Zap/g1/${EXTEN:1},90,r)
exten = _9X.,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_9X.,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(
num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = _9X.,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _9X.,4,Dial(Zap/g1/${EXTEN},90,r)
exten = _9X.,5,Busy
exten = 000,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
000,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(n
um)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN})
exten = 000,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = 000,4,Dial(Zap/g1/000,60,r)
exten = 9000,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
9000,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(
num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = 9000,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = 9000,4,Dial(Zap/g1/000,60,r)
[out-voip]
exten = _902X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _903X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _905X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _906X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _908X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _954X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _955X.,1,Dial(SIP/acevoip/${EXTEN:1})
[from-acevoip]
include = dialstring
exten = 073...,1,Answer
exten = 073...,2,Dial(Zap/g1/100,60,tn)
exten = _073.XX,1,Answer
exten = _073.XX,2,System(mkdir
/mnt/data/Recording/${SIP_HEADER(TO):12:3})
exten =
_073.XX,3,Set(CALLFILENAME=/mnt/data/Recording/${SIP_HEADER(TO):12:3
}/${SIP_HEADER(TO):12:3}-Received-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-$
{CALLERID(num)})
exten = _073.XX,4,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _073.XX,5,Dial(SIP/${SIP_HEADER(TO):12:3},60,tn)
exten = _073.XX,6,Voicemail(${SIP_HEADER(TO):12:3}u)
exten = _073.XX,7,Hangup
exten = _073.XX,106,Voicemail(${SIP_HEADER(TO):12:3}u)
exten = _073.XX,107,Hangup
include = dial-sip
include = dial-e100p
[from-sip]
include = dialstring
include = dial-sip
include = out-voip
include = dial-e100p
[dial-sip]
exten = 600,1,Dial(Zap/g1/100,60,tr)
exten = 9600,1,Dial(Zap/g1/100,60,tr)
exten = _6XX,1,SetMusicOnHold(random)
exten = _6XX,2,System(mkdir /mnt/data/Recording/${EXTEN})
exten =
_6XX,3,Set(CALLFILENAME=/mnt/data/Recording/${EXTEN}/${EXTEN}-Received-$
{STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${CALLERID(num)}.wav49)
exten = _6XX,4,MixMonitor(${CALLFILENAME}|v(0)V(0))
exten = _6XX,5,Dial(SIP/${EXTEN},45,Ttr)
exten = _6XX,6,Voicemail(u${EXTEN})
exten = _6XX,7,Hangup
exten = _6XX,106,Voicemail(b${EXTEN})
exten = _6XX,107,Hangup
exten = _96XX,1,SetMusicOnHold(random)
exten =
[Asterisk-Users] Asterisk receiving call from Panasonic TDA extension issue
Sorry someone screwing with permissions on my server bounced the 2 days worth of email after I posted this, any and all those lovely people who replied with suggestions from my post could you sent them again :-) James -Original Message- From: James Bean On Behalf Of Asterisk Mailing List Sent: Tuesday, 30 May 2006 12:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Asterisk receiving call from Panasonic TDA extension issue Asterisk, Zap and Libpri version from Asterisk SVN-branch-1.2-r27093 Error:- -- Accepting overlap call from '123' to '6' on channel 0/31, span 1 -- Starting simple switch on 'Zap/31-1' -- Hungup 'Zap/31-1' Primary Rate E1 30 trunks connecting between Asterisk and TDA200 Pansonic TDA200 has 1XX extensions Asterisk is setup with 6XX extensions If Asterisk calls a 1XX its not an issue, when 1XX calls Asterisk it looks like the phone system is dialing the digitals individually instead of at once so Asterisk is receiving the first 6 going I don't know 6 before it receives the rest of the digits from the TDA. Any clues as to if its possible to have asterisk wait for the rest of the digits, a wait of sorts, or I have to figure out how to make the TDA do it? James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk receiving call from Panasonic TDA extension issue
Asterisk, Zap and Libpri version from Asterisk SVN-branch-1.2-r27093 Error:- -- Accepting overlap call from '123' to '6' on channel 0/31, span 1 -- Starting simple switch on 'Zap/31-1' -- Hungup 'Zap/31-1' Primary Rate E1 30 trunks connecting between Asterisk and TDA200 Pansonic TDA200 has 1XX extensions Asterisk is setup with 6XX extensions If Asterisk calls a 1XX its not an issue, when 1XX calls Asterisk it looks like the phone system is dialing the digitals individually instead of at once so Asterisk is receiving the first 6 going I don't know 6 before it receives the rest of the digits from the TDA. Any clues as to if its possible to have asterisk wait for the rest of the digits, a wait of sorts, or I have to figure out how to make the TDA do it? James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line Dropouts on E405P
Hi,
We have a Ericsson BP250 Phone system setup witht
he following configuration
Telco - Asterisk E405P -
BP250
The system seem to work perfectly on 1.0.9 for a
very long time but there is some functionality we wanted to take advantage of in
the 1.2 version branch so we upgraded.
Currently running
Asterisk 1.2.4
Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade
next weekend)
Libpri 1.2.2
The problem we are getting is wierd but
:-
Sorry about the timings looking wierd but you have
to allow a fudge factor of anywhere upto 12 hours when dealing with reports from
on-site personel.
* Wednesday ~ 9.30am All calls drops for 1 second,
then back online, then ~5 minutes later, same thing* Thursday ~
10.30am All calls drops for 1 second, then back online, then ~5 minutes later,
same thing
* Friday ~ 11.30am All calls drops for 1 second,
then back online, then ~5 minutes later, same thing
Just before it drops out the calls sound a little fuzzy.
There is no warning messages on console.Error log (which seem to
correspond to drops outs):Feb 17 11:30:08 WARNING[2566] chan_zap.c: No
D-channels available! UsingPrimary channel 47 as D-channel
anyway!Feb 17 11:36:12 WARNING[2565] chan_zap.c: No D-channels
available! UsingPrimary channel 16 as D-channel anyway!
D-Channel 47 relates to thesocket which is connected to the BP250,
D-Channel 16 relates to the socket connected to the telco.
I really don't want to have to drop back to 1.0.9 if i can avoid it.
Log files and settings :-
Logger.conffull =
notice,warning,errorZaptel.confspan=1,1,0,ccs,hdb3,crc4bchan=1-15dchan=16bchan=17-31span=2,0,0,ccs,hdb3,crc4bchan=32-46dchan=47bchan=48-62span=3,0,0,ccs,hdb3,crc4bchan=63-77dchan=78bchan=79-93span=4,0,0,ccs,hdb3,crc4bchan=94-108dchan=109bchan=110-124
Zapata.conf
[channels]context=defaultmusiconhold=defaultswitchtype=euroisdnusecallerid=yescidsignalling=v23cidstart=polarityhidecallerid=nocallwaiting=nousecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesechotraining=800rxgain=0.0txgain=0.0
group=1context=te405p-intelstrapridialplan=localsignalling=pri_cpe;overlapdial=yescallerid=asreceivedchannel=1-15,
17-31group=4context=te405p-frombp250pridialplan=localsignalling=pri_netoverlapdial=yescallerid=asreceivedchannel=32-46,
48-62
Extensions.conf (Sorry for it being so large, most of the rest it of is in
other files)
[default]exten = s,1,Dial(SIP/5552,45,t)
[dialstring]
exten = i,1,Playback(invalid)exten = i,2,Hangupexten =
t,1,Hangup
[atp-out]
exten = _8X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]);exten =
_8X.,1,dial(SIP/${EXTEN:[EMAIL PROTECTED],30)exten =
_8X.,2,Congestion
exten =
_9X.,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN:1})exten
= _9X.,2,Congestionexten = _9X.,3,Hangup
[from-callpacket]
exten = 17025541498,1,Answerexten =
17025541498,2,Dial(SIP/557)exten = 17025541498,3,Hangup
[atp-in]
exten = 30182849,1,SetMusicOnHold(record)exten =
30182849,2,Dial(SIP/551,45,t)exten = 30182849,3,Voicemail,u551exten
= 30182849,103,Voicemail,b551
exten = s,1,Dial(SIP/3332,45,t)
[te405p-frombp250]
exten =
_321X.,1,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN:3})
include = to-sipinclude = parkedcallsinclude =
record-transferinclude = atp-outinclude = voicerecinclude
= lm1_functionsinclude = te405p-outtelstra
[te405p-tobp250]
#include extensions_te405p-tobp250.conf
[te405p-intelstra]
#include extensions_te405p-intelstra.confinclude = to-sip
[te405p-outtelstra]
#include extensions_te405p-outtelstra.confinclude =
dialstring
include = js_play_ael
[from-sip]exten =
555,1,dial(SIP/username:[EMAIL PROTECTED]/0732822922)
exten = 881,1,Dial(Zap/G4/38165912)exten =
982,1,Dial(Zap/G4/38166400)exten = 983,1,Dial(Zap/G4/38105000)exten
= 984,1,Dial(Zap/G4/5483)exten =
985,1,Dial(Zap/G4/5912)exten = 986,1,Dial(Zap/G4/5760)exten
= 987,1,Dial(Zap/G4/5765)exten =
988,1,Dial(Zap/G4/1006)exten = 989,1,Dial(Zap/G4/5947)exten
= 55,1,Dial(Zap/G1/0423813901)
exten = s,1,Dial(SIP/3332,45,t)
include = atp-outinclude = lm1_functionsinclude =
from-callpacketinclude = to-sipinclude =
te405p-tobp250include = te405p-outtelstrainclude =
record-transferinclude = parkedcallsinclude = voicerec
[record-transfer]
exten = _32XX,1,SetVar(DDATE=${TIMESTAMP})exten =
_32XX,2,SetVar(CALLFILENAME=/mnt/asterisk/pub/newbiz/${DDATE:0:8}/${EXTEN:1}/${EXTEN:1}-${TIMESTAMP})exten
= _32XX,3,Monitor(gsm,${CALLFILENAME},m)exten =
_32XX,4,Dial(ZAP/g4/${EXTEN:1})exten = _32XX,5,Congestionexten
= _32XX,105,Congestion
exten =
_34XX,1,SetVar(CALLFILENAME=/mnt/asterisk/5xxx/CallTo-${EXTEN:1}-${TIMESTAMP})exten
= _34XX,2,Monitor(gsm,${CALLFILENAME},m)exten =
_34XX,3,Dial(ZAP/g4/${EXTEN:1})exten = _34XX,4,Congestionexten
= _34XX,104,Congestion
exten =
_399X.,1,Dial(IAX2/username:[EMAIL PROTECTED]/0011${EXTEN:3})exten
= _399X.,2,Congestionexten = _399X.,3,Hangup
[voicerec]
exten = 381,1,Festival('Please record your
RE: [Asterisk-Users] Line Dropouts on E405P
I did some testing - More
Information - Hope this helps...
Next thing to try is to maybe move the port
that the Asterisk - BP250 (Group 1/D-Channel 16) resides on and see if that
makes a difference.
If I call 30 numbers from Asterisk --
BP250
only 28 connect and get the following
in the log file:Feb 19 08:46:22 NOTICE[13902] app_dial.c: Unable to
create channel of type'Zap' (cause 34 - Circuit/channel congestion)Feb
19 08:46:22 NOTICE[13902] app_dial.c: Unable to create channel of type'Zap'
(cause 34 - Circuit/channel
congestion)
If I call 15 extensions
via
Asterisk - Telstra - Asterisk
- BP250
I get the following in the log
file:Feb 19 08:51:41 WARNING[2565] chan_zap.c: Ring requested on channel
0/15already in use on span 1. Hanging up owner.Feb 19 08:51:42
WARNING[2565] chan_zap.c: Ring requested on channel 0/14already in use on
span 1. Hanging up owner.Feb 19 08:51:42 WARNING[2565] chan_zap.c:
Ring requested on channel 0/13already in use on span 1. Hanging up
owner.Feb 19 08:51:42 WARNING[2565] chan_zap.c: Got restart ack on channel
0/8span 1 with owner
As I dial the 15 as above I get the
following in the CLI:asterisk1*CLI -- Executing
Dial("SIP/3332-c760","Zap/g1/38165901Zap/g1/38165902Zap/g1/38165903Zap/g1/38165904Zap/g1/38165905Zap/g1/38165906Zap/g1/38165907Zap/g1/38165908Zap/g1/38165909Zap/g1/38165910Zap/g1/38165911Zap/g1/38165912Zap/g1/38165913Zap/g1/38165914Zap/g1/38165915")
in new stack -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165901 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165902 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165903 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165904 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165905 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165906 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165907 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165908 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165909 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165910 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165911 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165912 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165913 --
Requested transfer capability: 0x00 - SPEECH -- Called
g1/38165914 -- Requested transfer capability: 0x00 -
SPEECH -- Called g1/38165915!! Got reject for frame
77, retransmitting frame 77 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 78 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 79 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 80 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 81 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 82 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 83 now, updating n_r!!! Got reject for frame 77,
retransmitting frame 84 now, updating n_r! -- Zap/3-1 is
proceeding passing it to SIP/3332-c760 -- Zap/2-1 is
proceeding passing it to SIP/3332-c760 -- Zap/1-1 is
proceeding passing it to SIP/3332-c760 -- Zap/7-1 is
proceeding passing it to SIP/3332-c760 -- Zap/6-1 is
proceeding passing it to SIP/3332-c760 -- Zap/5-1 is
proceeding passing it to SIP/3332-c760 -- Zap/4-1 is
proceeding passing it to SIP/3332-c760 -- Channel 0/12,
span 1 got hangup -- Forcing restart of channel 0/12 on
span 1 since channel reported inuse -- Zap/11-1 is
proceeding passing it to SIP/3332-c760 -- Zap/10-1 is
proceeding passing it to SIP/3332-c760 -- Hungup
'Zap/12-1'!! Got reject for frame 86, retransmitting frame 86 now, updating
n_r!!! Got reject for frame 86, retransmitting frame 87 now, updating
n_r!!! Got reject for frame 86, retransmitting frame 88 now, updating
n_r! -- Zap/9-1 is proceeding passing it to
SIP/3332-c760 -- Accepting call from '738166400' to
'38165903' on channel 0/19, span 1 -- Executing
SetMusicOnHold("Zap/19-1", "record") in new stack --
Zap/8-1 is proceeding passing it to SIP/3332-c760 --
Executing Dial("Zap/19-1", "Zap/g4/38165903|6000|t") in new
stack -- Requested transfer capability: 0x00 -
SPEECH -- Called g4/38165903 --
Accepting call from '738166400' to '38165902' on channel 0/22, span
1 -- Executing SetMusicOnHold("Zap/22-1", "record") in new
stack -- Executing Dial("Zap/22-1",
"Zap/g4/38165902|6000|t") in new stack -- Requested
transfer capability: 0x00 - SPEECH -- Called
g4/38165902 -- Zap/32-1 is proceeding passing it to
Zap/19-1 -- Accepting call from '738166400' to '38165901'
on channel 0/17, span 1 -- Executing
SetMusicOnHold("Zap/17-1", "record") in new stack --
Executing Dial("Zap/17-1", "Zap/g4/38165901|6000|t") in new
stack
Re: [Asterisk-Users] chan_bluetooth and Ericcson T68 problem
Hello Enky, We have encountered similar problems with various Ericsson Nokia phones. We couldn't get the channel driver to work 100%. However, we cannot actually tell whether it was our mistake or whether there was a problem with the channel driver. We tried to contact the driver's maintainer/creator but no luck... If you manage to find a solution for this problem we'd also be interested to know about it. Best regards, Vlasis. Enky wrote: Hi, I have read many pages and tried many things, but without any success. I have paired my ERICCSON T68 with the Asterisk PC. The Asterisk version is “Asterisk CVS-v1-0-11/19/05-14:52:52”. The chan_bluetooth is the last release, downloaded from “http://www.crazygreek.co.uk/data/pages/chan_bluetooth/latest.tar.gz”. It is all OK. I can dial from the Asterisk a number. The T68 dials it, but when the called party picks the phone and the call goes connected there is no any audio! Neither from or to the Asterisk. Here are a short logs: This is the initial log, when I start the Asterisk and it connects the T68. It seems OK: ---cut--- Asterisk Ready. *CLI Nov 19 15:15:45 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2041 try_connect: Initialised bluetooth link to device T68 [AG]T68 AT+BRSF=23 [AG]T68 ERROR [AG]T68 AT+CIND=? [AG]T68 +CIND: (battchg,(0-5)),(signal,(0-5)),(batterywarning,(0-1)),(chargerconnected,(0-1)),(service,(0-1)),(sounder,(0-1)),(message,(0-1)),(call,(0-1)),(roam,(0-1)),(smsfull,(0-1)) [AG]T68 OK [AG]T68 AT+CIND? Nov 19 15:15:46 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:417 set_cind: Audio Gateway T68 got signal [AG]T68 +CIND: 5,5,0,1,1,0,0,0,0,0 [AG]T68 OK [AG]T68 AT+CMER=3,0,0,1 [AG]T68 OK [AG]T68 AT+CLIP=1 [AG]T68 OK [AG]T68 AT+CGMI=? [AG]T68 OK [AG]T68 AT+CGMI [AG]T68 ERICSSON [AG]T68 OK ---cut--- This is when I dial a number. It seems OK too, but no audio when connects: ---cut--- -- Executing Dial(SIP/222-3885, BLT/T68/123|60) in new stack [AG]T68 ATD123; -- Called T68 [AG]T68 OK [AG]T68 +CIEV: 8,1 -- BLT/T68 answered SIP/222-3885 [AG]T68 +CIEV: 2,4 [AG]T68 +CIEV: 2,5 ---cut--- And this is when I interrupt the dialed call: ---cut--- [AG]T68 AT+CHUP == Spawn extension (default, 2002, 1) exited non-zero on 'SIP/222-3885' [AG]T68 OK Nov 19 15:18:06 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2493 rd_close: Device T68 disconnected, scheduled reconnect in 5 seconds: Connection reset by peer (errno 104) Nov 19 15:18:11 NOTICE[11068]: /usr/src/asterisk/chan_bluetooth/chan_bluetooth.c:2041 try_connect: Initialised bluetooth link to device T68 [AG]T68 AT+BRSF=23 [AG]T68 ERROR [AG]T68 AT+CIND=? [AG]T68 +CIND: (battchg,(0-5)),(signal,(0-5)),(batterywarning,(0-1)),(chargerconnected,(0-1)),(service,(0-1)),(sounder,(0-1)),(message,(0-1)),(call,(0-1)),(roam,(0-1)),(smsfull,(0-1)) [AG]T68 OK [AG]T68 AT+CIND? [AG]T68 +CIND: 5,5,0,1,1,0,0,0,0,0 [AG]T68 OK [AG]T68 AT+CMER=3,0,0,1 [AG]T68 OK [AG]T68 AT+CLIP=1 [AG]T68 OK [AG]T68 AT+CGMI=? [AG]T68 OK [AG]T68 AT+CGMI [AG]T68 ERICSSON [AG]T68 OK ---cut--- Please someone to help me :) Thank you in advance! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 question
Angelito Manansala wrote: yes On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote: Hi all, for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp go through asterisk ) thanks in advance best regards! Hello, As far as I know Asterisk cannot disentangle RTP from signaling in either SIP or H323 at least until now. I'd also be interested to know if this option is available now in case I've missed something... Best regards, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon Diva Server query
Avi Miller wrote: Hello gurus! I've given up on crappy passive ISDN cards and am heading into the wild world of real, Active Super Dooper Server boards. I have a choice of two Eicon Diva Server cards: Eicon Diva Server 4BRI Eicon Diva Server V-4BRI Hello, We've been using an Eicon Diva Server 4BRI with a RH 9 installation (kernel 2.4.20-8). It works great in both TE and NT mode. I assume that it will work equally great with a 2.6 kernel... Best regard, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)
If anyone is interested I'm (slowly) developing a GPL'd Java applet that works as an IAX softphone. I should have a test version out at the end of the week for a limited number of testers. Tim. http://www.westhawk.co.uk/ Hello Tim, We'd be interested to test the client... Best regards, Vlasis Hatzistavrou Technical Director CEO Kinetix Tele.com Hellas Ltd. Monastiriou 9 Enotikon 546 27 Thessaloniki Greece Tel.: +302310556134 Fax: +302310556134 (ext. 0) GSM: +306977835653 e-mail: [EMAIL PROTECTED] http://www.kinetix.gr ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp - fax is just blank pages
pbo 808 wrote: I've done quite a bit of googling and haven't found a solution to my problem. I've got the Digium dev kit (wctdm11b) set up and working. I've compiled spandsp and can receieve faxes from eFax (www.efax.com) but the pages are blank. The page count is correct, in that if I fax a two page document, my tiff file has two pages, but they are white blank pages. I found one similar post here http://lists.digium.com/pipermail/asterisk-users/2005-April/103069.html, but haven't seen a solution. Any ideas? ___ Hello, I have noticed the same problem in my tests with spandsp. I think it has to do with the format of the tiff file, but I couldn't find the reason... I hope that someone in this list who has solved this problem can share the solution with us. Best regards, Vlasis Hatzistavrou. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
