Re: [asterisk-users] Asterisk 1.4 Fax
Well that answers that question. I see that t38modem provides an H232 modem - is this unsuitable for HylaFAX's purpose? (ignoring the fact that it requires a kernel recompile on most newer distros.) Steve Underwood wrote: Rob Hillis wrote: Last time I heard IAXModem didn't support T.38 because the IAX2 protocol didn't support T.38 - whether that's still the case or not, I don't know. There are actually two reasons. One is that T.38 over IAX is not defined. The other is the current T.38 termination support in spandsp is only for the full FAX machine it contains. T.38 termination to the class 1 FAX modem (T.31) interface for HylaFAX is a work in progress. When that is done, I hope we will have a sipmodem to replace iaxmodem, offering bother audio and T.38 to HylaFAX functionality. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Hi Rob, Rob Hillis wrote: Well that answers that question. I see that t38modem provides an H232 modem - is this unsuitable for HylaFAX's purpose? (ignoring the fact that it requires a kernel recompile on most newer distros.) Steve Underwood wrote: Rob Hillis wrote: Last time I heard IAXModem didn't support T.38 because the IAX2 protocol didn't support T.38 - whether that's still the case or not, I don't know. There are actually two reasons. One is that T.38 over IAX is not defined. The other is the current T.38 termination support in spandsp is only for the full FAX machine it contains. T.38 termination to the class 1 FAX modem (T.31) interface for HylaFAX is a work in progress. When that is done, I hope we will have a sipmodem to replace iaxmodem, offering bother audio and T.38 to HylaFAX functionality. Steve The most recent versions of t38modem can apparently provide both a SIP and H.323 T.38 to class 1 FAX modem interface for HylaFAX. What it cannot provide is an audio FAX interface. The sipmodem code I am working on will integrate audio and T.38 FAX processing in a single SIP entity. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Polycom Soundpoint IP 320 Hardphone
Unfortunately there is only one port, clearly labelled handset On 31/12/2007, at 11:34 PM, dave cantera wrote: glenn, check your handset cord... it might be plugged into the wrong port in the back of the phone. perhaps the headset jack... daveC Glenn Gillen wrote: Hey all, I've setup my asterisk install on a CentOS5 server, I've got a few IAX2 and SIP softphone clients connected on the same subnet and at least 1 external IAX2 softphone. However I'm having some difficulty getting the Polycom hardphone to function correctly. Watching the logs and debug trace it: - Registers correctly - Is able to make calls to other peers However it is not able to answer calls made to it. That is, the handset actually rings, but I've no way to answer it. The answer soft key, picking up the phone, etc. all have no effect. And I'm at a loss as to what setting should be altered to fix it. Any ideas? Possibly a tangent, but also affecting this handset, is that trying to dial out over an external SIP trunk fails on the first attempt. But calling an internal peer and then trying a second time makes it mysteriously work. Any help greatly appreciated, ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r 8000 -c 1 -s -w test_wav_out.wav resample -ql But it seems like I'm missing the codec or something: === -- Executing [EMAIL PROTECTED]:2] Playback(SIP/2000-0871d000, /usr/local/lib/asterisk/test_wav_out.wav) in new stack WARNING[37390]: file.c:563 ast_openstream_full: File /usr/local/lib/asterisk/test_wav_out.wav does not exist in any format WARNING[37390]: file.c:866 ast_streamfile: Unable to open /usr/local/lib/asterisk/test_wav_out.wav (format 0x4 (ulaw)): No such file or directory === Here's what core show file formats says: === Format Name Extensions gsmwav49 WAV|wav49 slin wavwav adpcm voxvox slin slnsln|raw g722 g722 g722 ulaw au au alaw alaw alaw|al ulaw pcmpcm|ulaw|ul|mu ilbc iLBC ilbc h264 h264 h264 h263 h263 h263 gsmgsmgsm g729 g729 g729 g726 g726-16g726-16 g726 g726-24g726-24 g726 g726-32g726-32 g726 g726-40g726-40 g723 g723sf g723|g723sf 18 file formats registered. === Am I missing something in the configuration files, or maybe I'm missing some module? Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
On Jan 1, 2008 3:36 PM, Vincent [EMAIL PROTECTED] wrote: Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r 8000 -c 1 -s -w test_wav_out.wav resample -ql But it seems like I'm missing the codec or something: === -- Executing [EMAIL PROTECTED]:2] Playback(SIP/2000-0871d000, /usr/local/lib/asterisk/test_wav_out.wav) in new stack WARNING[37390]: file.c:563 ast_openstream_full: File /usr/local/lib/asterisk/test_wav_out.wav does not exist in any format WARNING[37390]: file.c:866 ast_streamfile: Unable to open /usr/local/lib/asterisk/test_wav_out.wav (format 0x4 (ulaw)): No such file or directory === Happy New Year! It seems from the console output that you have specified the extension of the filename in your dialplan. That doesn't work with asterisk. All you need to do is specify the name of the file you want to play without the extension like s,2,Playback(/usr/local/lib/asterisk/test_wav_out) And asterisk will automatically pickup the file that it can play with any asterisk supported format from the specified path. -- Godson Gera, http://godson.in Asterisk India ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Polycom Soundpoint IP 320 Hardphone
Glenn Gillen wrote: Unfortunately there is only one port, clearly labelled handset On 31/12/2007, at 11:34 PM, dave cantera wrote: glenn, check your handset cord... it might be plugged into the wrong port in the back of the phone. perhaps the headset jack... daveC Push the cord all the way into the handset. I've seen some Polycom handsets that look like they are plugged in, but in reality, the end of the cord that plugs into the handset needs to go in farther. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
On Tue, 1 Jan 2008 17:23:29 +0530, Godson Gera [EMAIL PROTECTED] wrote: s,2,Playback(/usr/local/lib/asterisk/test_wav_out) And asterisk will automatically pickup the file that it can play with any asterisk supported format from the specified path. OK. Is there a way to tell Asterisk which codec to use so it doesn't try figuring out the file format used? Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
Vincent wrote: On Tue, 1 Jan 2008 17:23:29 +0530, Godson Gera [EMAIL PROTECTED] wrote: s,2,Playback(/usr/local/lib/asterisk/test_wav_out) And asterisk will automatically pickup the file that it can play with any asterisk supported format from the specified path. OK. Is there a way to tell Asterisk which codec to use so it doesn't try figuring out the file format used? Thanks. The codec is specified (for a sip device) in sip.conf, like this: [general] disallow=all allow=ulaw allow=alaw allow=gsm And you know that you can convert the files to every codec format that is in use then will the cpu load be minimalized ! To convert between different codec formats can you use the asterisk CLI command: file convert file_in.format file_out.format To convert from a shell script can you do like this: #!/bin/bash # Converts a audio file from alaw to a ulaw rasterisk -x file convert /tmp/file_in.alaw /tmp/file_out.ulaw More examples: The old way: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk I will try to update this page with convert. As a final touch, I have heard that sln should be the prefered format where you dont have the same format as the codec used in a channel. Asterisk native format is sln /Mats ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
On Jan 1, 2008 3:36 PM, Vincent [EMAIL PROTECTED] wrote: Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk Asterisk automatically takes care of saving CPU issue as it picks the file that have less translation cost (in other words it picks the file that gives the best CPU performance based on call situations like in which codec format the call is bridged ). That way you don't have to worry about specifying particular format moreover there is no provision to do that in Playback application. You can see translation costs by typing the following in console. core show translation -- Godson Gera, http://godsongera.blogspot.com Asterisk India Developer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + SIP + cisco phone confrance problem
Dear all I have cisco phone 7974 i have useing SIP protocol to register phone on Asterisk and it is working fine but i have one problem when how do i use confranceing between 2 party i am not talking about meetme confrance i am taking about phone confranceing like press flash key and take other caller PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org - Looking for last minute shopping deals? Find them fast with Yahoo! Search.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
I'm not looking at T.38 , at this time its terminating a SIP trunk with multiple DID's for fax. I'm using this configuration with linksys PAP ATA and satisfied with results. I'm looking at removing these ATA 's and using Asterisk ( or giving it a try ) for terminating fax. Last time I heard IAXModem didn't support T.38 because the IAX2 protocol didn't support T.38 - whether that's still the case or not, I don't know. There are actually two reasons. One is that T.38 over IAX is not defined. The other is the current T.38 termination support in spandsp is only for the full FAX machine it contains. T.38 termination to the class 1 FAX modem (T.31) interface for HylaFAX is a work in progress. When that is done, I hope we will have a sipmodem to replace iaxmodem, offering bother audio and T.38 to HylaFAX functionality. Steve The most recent versions of t38modem can apparently provide both a SIP and H.323 T.38 to class 1 FAX modem interface for HylaFAX. What it cannot provide is an audio FAX interface. The sipmodem code I am working on will integrate audio and T.38 FAX processing in a single SIP entity. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
vincent,
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in the
current directory...
creates a logfile, although trivial...
daveC
#!/bin/sh
#
# convert-all.sh
#
# convert all *.wav files to .gsm .au formats
#
if [ "null${1}" == "null" ]
then
FILE_LIST=`ls *.wav`
else
FILE_LIST=`ls ${1}*.wav`
fi
LOG="./log_convert.log"
echo "=== "
${LOG}
echo " started at `date` " ${LOG}
echo " Removing all current .gsm files..."
rm -f *.gsm
for FNAME in ${FILE_LIST}
do
echo " --- - "
echo " " ${LOG}
echo " Processing ${FNAME}... "
echo " Processing ${FNAME}... " ${LOG}
BASEFNAME=`echo ${FNAME} | awk '{print substr($0,1,length($0)-4)}'`
echo " making ${BASEFNAME}.gsm... "
echo " making ${BASEFNAME}.gsm... " ${LOG}
#sox -q -V -c 1 ${FNAME} -r 8000 -c 1 -w ${BASEFNAME}.gsm resample
-ql 2${LOG}
sox -q -V ${FNAME} -r 8000 -c 1 ${BASEFNAME}.gsm resample -ql
2${LOG}
echo " " ${LOG}
echo " making ${BASEFNAME}.au... "
echo " making ${BASEFNAME}.au... " ${LOG}
sox -q -V ${FNAME} -t au -r 8000 -c 1 -w ${BASEFNAME}.au resample
-ql 2${LOG}
done
Vincent wrote:
Hello
Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0
and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd
like to play PCM WAV files instead of eg. GSM. Per...
www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
... I recorded a sample of my voice using XP's Sound Recorder, then
ran the following :
sox test_wav.wav -r 8000 -c 1 -s -w test_wav_out.wav resample -ql
But it seems like I'm missing the codec or something:
===
-- Executing [EMAIL PROTECTED]:2] Playback("SIP/2000-0871d000",
"/usr/local/lib/asterisk/test_wav_out.wav") in new stack
WARNING[37390]: file.c:563 ast_openstream_full: File
/usr/local/lib/asterisk/test_wav_out.wav does not exist in any format
WARNING[37390]: file.c:866 ast_streamfile: Unable to open
/usr/local/lib/asterisk/test_wav_out.wav (format 0x4 (ulaw)): No such
file or directory
===
Here's what "core show file formats" says:
===
Format Name Extensions
gsmwav49 WAV|wav49
slin wavwav
adpcm voxvox
slin slnsln|raw
g722 g722 g722
ulaw au au
alaw alaw alaw|al
ulaw pcmpcm|ulaw|ul|mu
ilbc iLBC ilbc
h264 h264 h264
h263 h263 h263
gsmgsmgsm
g729 g729 g729
g726 g726-16g726-16
g726 g726-24g726-24
g726 g726-32g726-32
g726 g726-40g726-40
g723 g723sf g723|g723sf
18 file formats registered.
===
Am I missing something in the configuration files, or maybe I'm
missing some module?
Thank you.
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[asterisk-users] (no subject)
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Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
Thanks nice script.
But why au files in addition to gsm?
- Original Message -
From: dave cantera
To: Asterisk Users Mailing List - Non-Commercial Discussion ; [EMAIL
PROTECTED]
Sent: Tuesday, January 01, 2008 11:27 AM
Subject: Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
vincent,
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in the current
directory...
creates a logfile, although trivial...
daveC
#!/bin/sh
#
#convert-all.sh
#
#convert all *.wav files to .gsm .au formats
#
if [ null${1} == null ]
then
FILE_LIST=`ls *.wav`
else
FILE_LIST=`ls ${1}*.wav`
fi
LOG=./log_convert.log
echo === ${LOG}
echo started at `date` ${LOG}
echo Removing all current .gsm files...
rm -f *.gsm
for FNAME in ${FILE_LIST}
do
echo --- -
echo ${LOG}
echo Processing ${FNAME}...
echo Processing ${FNAME}... ${LOG}
BASEFNAME=`echo ${FNAME} | awk '{print substr($0,1,length($0)-4)}'`
echo making ${BASEFNAME}.gsm...
echo making ${BASEFNAME}.gsm... ${LOG}
#sox -q -V -c 1 ${FNAME} -r 8000 -c 1 -w ${BASEFNAME}.gsm resample -ql
2${LOG}
sox -q -V ${FNAME} -r 8000 -c 1 ${BASEFNAME}.gsm resample -ql 2${LOG}
echo ${LOG}
echo making ${BASEFNAME}.au...
echo making ${BASEFNAME}.au... ${LOG}
sox -q -V ${FNAME} -t au -r 8000 -c 1 -w ${BASEFNAME}.au resample -ql
2${LOG}
done
Vincent wrote:
Hello
Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0
and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd
like to play PCM WAV files instead of eg. GSM. Per...
www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
... I recorded a sample of my voice using XP's Sound Recorder, then
ran the following :
sox test_wav.wav -r 8000 -c 1 -s -w test_wav_out.wav resample -ql
But it seems like I'm missing the codec or something:
===
-- Executing [EMAIL PROTECTED]:2] Playback(SIP/2000-0871d000,
/usr/local/lib/asterisk/test_wav_out.wav) in new stack
WARNING[37390]: file.c:563 ast_openstream_full: File
/usr/local/lib/asterisk/test_wav_out.wav does not exist in any format
WARNING[37390]: file.c:866 ast_streamfile: Unable to open
/usr/local/lib/asterisk/test_wav_out.wav (format 0x4 (ulaw)): No such
file or directory
===
Here's what core show file formats says:
===
Format Name Extensions
gsmwav49 WAV|wav49
slin wavwav
adpcm voxvox
slin slnsln|raw
g722 g722 g722
ulaw au au
alaw alaw alaw|al
ulaw pcmpcm|ulaw|ul|mu
ilbc iLBC ilbc
h264 h264 h264
h263 h263 h263
gsmgsmgsm
g729 g729 g729
g726 g726-16g726-16
g726 g726-24g726-24
g726 g726-32g726-32
g726 g726-40g726-40
g723 g723sf g723|g723sf
18 file formats registered.
===
Am I missing something in the configuration files, or maybe I'm
missing some module?
Thank you.
___
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--
My wife's sister is in California.
I should buy her a Videophone2008!
Truly, The Next Best Thing to Being There!
--
WorldWideVideoPhones.com
856.380.0894
--
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--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.516 / Virus Database: 269.17.13/1205 - Release Date: 12/31/2007
3:32 PM
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[asterisk-users] With rtcachefriends=yes, when do realtime changes take effect?
I asked this question last week and never got an answer. I also didn't find the answer in the wiki. I think it would be nice if asterisk would check the database again if the user re-registers, but it doesn't seem to do that. A periodic update would be ok too, but it doesn't seem to do that either. It seems like changes never happen until a reload.if that is the case then doesn't rtcachefriends completely defeat the purpose of realtime SIP users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
On Tue, Jan 01, 2008 at 11:27:54AM -0500, dave cantera wrote:
vincent,
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in the current
directory...
creates a logfile, although trivial...
daveC
#!/bin/sh
#
#convert-all.sh
#
#convert all *.wav files to .gsm .au formats
#
if [ null${1} == null ]
then
FILE_LIST=`ls *.wav`
else
FILE_LIST=`ls ${1}*.wav`
fi
LOG=./log_convert.log
echo === ${LOG}
echo started at `date` ${LOG}
echo Removing all current .gsm files...
rm -f *.gsm
# A note from the Useless Use of ls Committee:
for FNAME in $1*.wav
for FNAME in ${FILE_LIST}
do
echo --- -
echo ${LOG}
echo Processing ${FNAME}...
echo Processing ${FNAME}... ${LOG}
BASEFNAME=`echo ${FNAME} | awk '{print substr($0,1,length($0)-4)}'`
echo making ${BASEFNAME}.gsm...
echo making ${BASEFNAME}.gsm... ${LOG}
#sox -q -V -c 1 ${FNAME} -r 8000 -c 1 -w ${BASEFNAME}.gsm resample -ql
2${LOG}
sox -q -V ${FNAME} -r 8000 -c 1 ${BASEFNAME}.gsm resample -ql 2${LOG}
echo ${LOG}
echo making ${BASEFNAME}.au...
echo making ${BASEFNAME}.au... ${LOG}
sox -q -V ${FNAME} -t au -r 8000 -c 1 -w ${BASEFNAME}.au resample -ql 2$
{LOG}
done
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] With rtcachefriends=yes, when do realtime changes take effect?
Adam Moffett wrote: I asked this question last week and never got an answer. I also didn't find the answer in the wiki. I think it would be nice if asterisk would check the database again if the user re-registers, but it doesn't seem to do that. A periodic update would be ok too, but it doesn't seem to do that either. It seems like changes never happen until a reload.if that is the case then doesn't rtcachefriends completely defeat the purpose of realtime SIP users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users New entries take effect immediately, however changes require a sip reload. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init
Hi, Before I report a bug on http://bugs.digium.com, I would like to know if someone is seeing the same error message. Personally I am not using wctdm24xxp but other modules such as wcte12xp and wctdm. The latter modules load fine and are compiled with pci_register_driver as expected. The only module that seems to require the deprecated function pci_module_init is wctdm24xxp. Is this normal? Thanks, Vieri Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init
On Tue, Jan 01, 2008 at 10:24:24AM -0800, Vieri wrote: Hi, Before I report a bug on http://bugs.digium.com, I would like to know if someone is seeing the same error message. Personally I am not using wctdm24xxp but other modules such as wcte12xp and wctdm. The latter modules load fine and are compiled with pci_register_driver as expected. The only module that seems to require the deprecated function pci_module_init is wctdm24xxp. Is it a custom kernel that has no PCI support? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Check your extensions.conf On Jan 1, 2008 11:33 AM, lists65 [EMAIL PROTECTED] wrote: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init
--- Tzafrir Cohen [EMAIL PROTECTED] wrote: Is it a custom kernel that has no PCI support? It's a custom 2.6.22 with # grep -i pci /usr/src/linux/.config # Bus options (PCI, PCMCIA, EISA, MCA, ISA) CONFIG_PCI=y # CONFIG_PCI_GOBIOS is not set # CONFIG_PCI_GOMMCONFIG is not set # CONFIG_PCI_GODIRECT is not set CONFIG_PCI_GOANY=y CONFIG_PCI_BIOS=y CONFIG_PCI_DIRECT=y CONFIG_PCI_MMCONFIG=y CONFIG_PCIEPORTBUS=y CONFIG_PCIEAER=y # CONFIG_PCI_MSI is not set CONFIG_EISA_PCI_EISA=y # CONFIG_HOTPLUG_PCI is not set CONFIG_BLK_DEV_IDEPCI=y CONFIG_IDEPCI_SHARE_IRQ=y CONFIG_IDEPCI_PCIBUS_ORDER=y CONFIG_BLK_DEV_IDEDMA_PCI=y # CONFIG_PATA_CMD640_PCI is not set # CONFIG_IEEE1394_PCILYNX is not set CONFIG_NET_PCI=y CONFIG_NE2K_PCI=m CONFIG_TMSPCI=m CONFIG_PCI200SYN=m CONFIG_DSCC4_PCISYNC=y CONFIG_DSCC4_PCI_RST=y CONFIG_ISDN_DRV_AVMB1_B1PCI=m CONFIG_ISDN_DRV_AVMB1_B1PCIV4=y CONFIG_ISDN_DRV_AVMB1_T1PCI=m CONFIG_ISDN_DIVAS_BRIPCI=y CONFIG_ISDN_DIVAS_PRIPCI=y CONFIG_SERIO_PCIPS2=m CONFIG_SERIAL_8250_PCI=y Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan
ATT or Verizon. I think those are the only ILECs left, right? On Dec 31, 2007 9:26 AM, Steve Finkelstein [EMAIL PROTECTED] wrote: Senad, Mind if I ask who that provider is? Thanks. Sent from my iPhone On Dec 31, 2007, at 8:10 AM, Senad Jordanovic [EMAIL PROTECTED] wrote: Justin Case wrote: Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you. Yap I agree... but but for about $900 per month one could get T1 (24 channels) unlimited in/out as far I seen last time our providers rates. Senad On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any potential leads. Happy holidays! - sf ___ --Bandwidth and Colocation Provided by http://www.api- digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Then I suggest you prepare yourself for a lot of pain. Fax over the 'net without T.38 is almost guaranteed to not work. Al lists wrote: I'm not looking at T.38 , at this time its terminating a SIP trunk with multiple DID's for fax. I'm using this configuration with linksys PAP ATA and satisfied with results. I'm looking at removing these ATA 's and using Asterisk ( or giving it a try ) for terminating fax. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan
Andrew Joakimsen wrote: ATT or Verizon. I think those are the only ILECs left, right? Don't forget the company formerly known as US Worst Now Quest John Novack On Dec 31, 2007 9:26 AM, Steve Finkelstein [EMAIL PROTECTED] wrote: Senad, Mind if I ask who that provider is? Thanks. Sent from my iPhone On Dec 31, 2007, at 8:10 AM, Senad Jordanovic [EMAIL PROTECTED] wrote: Justin Case wrote: Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you. Yap I agree... but but for about $900 per month one could get T1 (24 channels) unlimited in/out as far I seen last time our providers rates. Senad On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any potential leads. Happy holidays! - sf ___ --Bandwidth and Colocation Provided by http://www.api- digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my co-pilot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One Way Delay in Audio Over Analog
What are you using for a PSTN gateway? From: Brian Alexander [mailto:[EMAIL PROTECTED] Sent: Monday, December 31, 2007 3:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] One Way Delay in Audio Over Analog I have been trying to track down the cause/fix for a problem and I am out of ideas... I am hoping one of you can point me in the right direction. The symptom is that when a calls is placed from an internal extension through an analog line to a number on the pstn the caller can hear the callee but the callee can not hear the caller for as long as ten seconds. The problem appears to happen fairly consistently on the same pstn numbers. However, I have not seen a common characteristic in those numbers. For example, one of them is a direct number to a cell phone and another is to a Verizon fiber-optic phone/data service. The problem does not seem to be related to the type of SIP phone being used by the caller - for example, we have tried both X-Lite and Polycom phones without a change in behavior. The problem does not appear to occur if the callee then calls into our system (at least the one time I was able to have this happen). Turning on or off echo cancellation and/or call progress does not seem to change the behavior. I will appreciate any ideas you have. I am certainly stumped. Thanks and Happy New Year! -Brian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Andrew Joakimsen wrote: Check your extensions.conf Hahahahaha! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
REALY?? Humm I have been doing this for over a year and we receive over 400 faxes a month! 8 iaxmodems with DID's from a real SIP provider. And this connection is used for ALL office traffic, mail, VPN, webmail, and DNS. NO echo and no voice quality issues. Now we do have a 12mb down 768k up connection. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis Sent: Tuesday, January 01, 2008 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 Fax Then I suggest you prepare yourself for a lot of pain. Fax over the 'net without T.38 is almost guaranteed to not work. Al lists wrote: I'm not looking at T.38 , at this time its terminating a SIP trunk with multiple DID's for fax. I'm using this configuration with linksys PAP ATA and satisfied with results. I'm looking at removing these ATA 's and using Asterisk ( or giving it a try ) for terminating fax. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk access Postgres for Realtime Configuration
Yes you can use res_conf_pgsql.so is present in asterisk 1.4 On Oct 7, 2006 1:22 AM, John Miloo [EMAIL PROTECTED] wrote: Hello Comunity, How can I get Asterisk realtime working with Postgres? (without ODBC)? Thanks John /doc/realtime.txt in Version 1.4 Beta2 Currently there are three realtime database drivers: * ODBC: Support for UnixODBC, integrated into Asterisk The UnixODBC subsystem supports many different databases, please check www.unixodbc.org for more information. * MySQL: Found in the asterisk-addons subversion repository on svn.digium.com * PostgreSQL: Native support for Postgres, integrated into Asterisk ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mehdi http://www.voz-ip.info ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Jonn R Taylor wrote: REALY?? Humm I have been doing this for over a year and we receive over 400 faxes a month! 8 iaxmodems with DID's from a real SIP provider. And this connection is used for ALL office traffic, mail, VPN, webmail, and DNS. NO echo and no voice quality issues. Now we do have a 12mb down 768k up connection. How often are you checking your HylaFAX+ Logs? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
I have it setup to email me any failed fax connections. Most of the faxes come from remote offices, distributors and customers. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, January 01, 2008 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 Fax Jonn R Taylor wrote: REALY?? Humm I have been doing this for over a year and we receive over 400 faxes a month! 8 iaxmodems with DID's from a real SIP provider. And this connection is used for ALL office traffic, mail, VPN, webmail, and DNS. NO echo and no voice quality issues. Now we do have a 12mb down 768k up connection. How often are you checking your HylaFAX+ Logs? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Jonn R Taylor wrote: I have it setup to email me any failed fax connections. Most of the faxes come from remote offices, distributors and customers. Same here, but HylaFAX won't send you any logs of attempts that haven't at least negotiated a fax transmission. Call comes in, tries to sync up several times and then hangs up. It gets logged, but doesn't get sent to the FaxMaster. You may want to check. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
On Tue, 2008-01-01 at 13:48 -0600, Jonn R Taylor wrote: REALY?? Humm I have been doing this for over a year and we receive over 400 faxes a month! 8 iaxmodems with DID's from a real SIP provider. And this connection is used for ALL office traffic, mail, VPN, webmail, and DNS. NO echo and no voice quality issues. Now we do have a 12mb down 768k up connection. Can you share more details about your implementation? what are you using for faxing? Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan
[Footers trimmed to protect my precious bandwidth. MY PRECIOUS!] Yes, but he didn't qualify it. You can get a T1 with unlimited minutes in the US -- as long as those minutes are local-only. Long distance is another matter, although most providers sell their voice T1s with a block of long distance minutes per month. It's certainly not unlimited, though. Also note that some people pay extra to have their local calling area cover virtually an entire state, which may seem like long distance to them. On Tuesday 01 January 2008 12:50:11 Andrew Joakimsen wrote: ATT or Verizon. I think those are the only ILECs left, right? On Dec 31, 2007 9:26 AM, Steve Finkelstein [EMAIL PROTECTED] wrote: Mind if I ask who that provider is? On Dec 31, 2007, at 8:10 AM, Senad Jordanovic [EMAIL PROTECTED] wrote: Justin Case wrote: Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you. Yap I agree... but but for about $900 per month one could get T1 (24 channels) unlimited in/out as far I seen last time our providers rates. On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
If I had ANY failed faxes I would here about it. Iaxmodem creates a log of its own, so when I get a connection that fails hylafax sends the failure to me. One of the things that I found is you need to add nojitterbuffer to the iaxmodem config file, only use g711, and you must have QOS enabled on your switches and/or a traffic shaper on your internet connection. I have a remote office that uses an IAX trunk and I can fax between these to offices over the internet. I have both app_txfax and app_rxfax also setup on asterisk and can use any of them. We also have 1 linksys ata that has a networked brother printer/fax and we can send faxes from it to any of the fax services on our network or any PSTN number. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, January 01, 2008 2:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 Fax Jonn R Taylor wrote: I have it setup to email me any failed fax connections. Most of the faxes come from remote offices, distributors and customers. Same here, but HylaFAX won't send you any logs of attempts that haven't at least negotiated a fax transmission. Call comes in, tries to sync up several times and then hangs up. It gets logged, but doesn't get sent to the FaxMaster. You may want to check. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Jonn R Taylor wrote: If I had ANY failed faxes I would here about it. Iaxmodem creates a log of its own, so when I get a connection that fails hylafax sends the failure to me. One of the things that I found is you need to add nojitterbuffer to the iaxmodem config file, Really? I'll have to do some testing, I've never tried since I've read you can't. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Jonn R Taylor wrote: One of the things that I found is you need to add nojitterbuffer to the iaxmodem config file The reason that you need the nojitterbuffer in the iaxmodem config file is because you're actually getting at least some jitter. IAXmodem's jitterbuffer simply fills-in gaps due to jitter with previously-heard audio samples. There is no way to recreate the missing audio. Filling-in the gaps with previous audio samples is effective in preventing premature carrier loss conditions, but it messes up the modems until real carrier loss does occur. It turns out that in most cases it's better to simply skip over the missing audio. The DSP seems to handle that quite gracefully. Thanks, Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
I have always said that if some one said it can't be done, they did not try hard enough. FYI... I love this. Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Tuesday, January 01, 2008 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 Fax Jonn R Taylor wrote: If I had ANY failed faxes I would here about it. Iaxmodem creates a log of its own, so when I get a connection that fails hylafax sends the failure to me. One of the things that I found is you need to add nojitterbuffer to the iaxmodem config file, Really? I'll have to do some testing, I've never tried since I've read you can't. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
That is correct. I found that out awhile ago with our internal fax. It would not connect, but the external faxes coming in over SIP worked. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Tuesday, January 01, 2008 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 Fax Jonn R Taylor wrote: One of the things that I found is you need to add nojitterbuffer to the iaxmodem config file The reason that you need the nojitterbuffer in the iaxmodem config file is because you're actually getting at least some jitter. IAXmodem's jitterbuffer simply fills-in gaps due to jitter with previously-heard audio samples. There is no way to recreate the missing audio. Filling-in the gaps with previous audio samples is effective in preventing premature carrier loss conditions, but it messes up the modems until real carrier loss does occur. It turns out that in most cases it's better to simply skip over the missing audio. The DSP seems to handle that quite gracefully. Thanks, Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
I'd say consider yourself very lucky. I know I did some testing here some time ago with faxing over VoIP. * One extension to another over G711a with both extensions on the same LAN - worked 95% of the time * One extension on my Asterisk server to an Extension on a friend's Asterisk server using G711a via IAX - 95% failure rate. Both of us awere on the same ISP and had ping times of ~40ms between us. However, in a live environment, I convert a PSTN call to a t.38 encoded call and can send the fax just about anywhere I damn well want (where the remote end supports t.38) with a 95% success rate. t.38 is the key to successful faxing over a VoIP network. Without it, you're begging for trouble. Doug Lytle wrote: Jonn R Taylor wrote: If I had ANY failed faxes I would here about it. Iaxmodem creates a log of its own, so when I get a connection that fails hylafax sends the failure to me. One of the things that I found is you need to add nojitterbuffer to the iaxmodem config file, Really? I'll have to do some testing, I've never tried since I've read you can't. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
NOT true and I have proven that for the last year. Jonn _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis Sent: Tuesday, January 01, 2008 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 Fax I'd say consider yourself very lucky. I know I did some testing here some time ago with faxing over VoIP. * One extension to another over G711a with both extensions on the same LAN - worked 95% of the time * One extension on my Asterisk server to an Extension on a friend's Asterisk server using G711a via IAX - 95% failure rate. Both of us awere on the same ISP and had ping times of ~40ms between us. However, in a live environment, I convert a PSTN call to a t.38 encoded call and can send the fax just about anywhere I damn well want (where the remote end supports t.38) with a 95% success rate. t.38 is the key to successful faxing over a VoIP network. Without it, you're begging for trouble. Doug Lytle wrote: Jonn R Taylor wrote: If I had ANY failed faxes I would here about it. Iaxmodem creates a log of its own, so when I get a connection that fails hylafax sends the failure to me. One of the things that I found is you need to add nojitterbuffer to the iaxmodem config file, Really? I'll have to do some testing, I've never tried since I've read you can't. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Jonn R Taylor wrote: FYI... I love this. Ben Franklin quote: I truly believe it. But, it being a Franklin quote is in some dispute. I like it all the same. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Jonn R Taylor wrote: I have always said that if some one said it can't be done, they did not try hard enough. FYI... I love this. Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. As the person behind the tools you are relying on, I can say you haven't tried hard at all. You are just lucky, and almost certainly just being very reliant on the majority of your FAXes using ECM mode, and retrying a lot. Trying hard for FAX over IP means implementing T.37, or at least T.38. These are engineered solutions, not pot luck. Your present arrangement assumes G.711 (not available a lot of the time), no signal manipulation in the system beyond your controls (getting rarer and rarer), a very crude network doing nothing to improve voice quality (should be getting rarer too), limited packet loss (which is truly pot luck over the internet, which you say you use), and a few other magic qualities. A number of people claim solid FAXing results across VoIP paths, like they've achieved some engineering breakthrough. The claims tend to evaporate under closer inspection. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Guys! what i was looking here was a simple hint/recommendation for installing IaxModem and Hylafax. Let me try it myself and see how feasible this solutions is. On Jan 1, 2008 5:02 PM, Steve Underwood [EMAIL PROTECTED] wrote: Jonn R Taylor wrote: I have always said that if some one said it can't be done, they did not try hard enough. FYI... I love this. Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. As the person behind the tools you are relying on, I can say you haven't tried hard at all. You are just lucky, and almost certainly just being very reliant on the majority of your FAXes using ECM mode, and retrying a lot. Trying hard for FAX over IP means implementing T.37, or at least T.38. These are engineered solutions, not pot luck. Your present arrangement assumes G.711 (not available a lot of the time), no signal manipulation in the system beyond your controls (getting rarer and rarer), a very crude network doing nothing to improve voice quality (should be getting rarer too), limited packet loss (which is truly pot luck over the internet, which you say you use), and a few other magic qualities. A number of people claim solid FAXing results across VoIP paths, like they've achieved some engineering breakthrough. The claims tend to evaporate under closer inspection. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
Steve, One of the main reasons that this works is controlling the data to and from the internet. I have spent the last 10 years building networks for ISP's. The key is getting the data from point a to point b in tact and in order. I did not get lucky as you put it. I am a network engineer and I know how to make networks work the way they need to. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Tuesday, January 01, 2008 6:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 Fax Jonn R Taylor wrote: I have always said that if some one said it can't be done, they did not try hard enough. FYI... I love this. Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. As the person behind the tools you are relying on, I can say you haven't tried hard at all. You are just lucky, and almost certainly just being very reliant on the majority of your FAXes using ECM mode, and retrying a lot. Trying hard for FAX over IP means implementing T.37, or at least T.38. These are engineered solutions, not pot luck. Your present arrangement assumes G.711 (not available a lot of the time), no signal manipulation in the system beyond your controls (getting rarer and rarer), a very crude network doing nothing to improve voice quality (should be getting rarer too), limited packet loss (which is truly pot luck over the internet, which you say you use), and a few other magic qualities. A number of people claim solid FAXing results across VoIP paths, like they've achieved some engineering breakthrough. The claims tend to evaporate under closer inspection. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How does Asterisk scale to 500-1000 phones?
Jesse, We have multiple installations of this scale and a few with far more concurrent call paths (250+). In our experience, Asterisk scales nicely to these levels as long as you are realistic about what you expect of the server. For instance, we rarely, if ever, convert signal to TDM. We instead use SIP dial tone from a tier-1 carrier. Also, if you expect any substantial amount of meetme conferences, you might want to consider running those on separate hardware. As the numbers go up, you can peel-apart your switch into functional duties such as two SIP switching servers, two voicemail servers, one conferencing server, etc. Just some ideas. Best of luck to you! Bryan M. Johns Shelton | Johns Office: 678.248.2637 FindMe: 678.229.1809 Support: [EMAIL PROTECTED] http://www.sheltonjohns.com On Dec 27, 2007, at 11:33 AM, Jesse Molina wrote: Anyone have opinions on how well Asterisk scales to 500-1000 stations, in regards to reliability, system performance, as well as ease of management? Assume relatively low call volume; let's say two public network PRIs (47 DS0s). -- # Jesse Molina # The Translational Genomics Research Institute # http://www.tgen.org # Mail = [EMAIL PROTECTED] # Desk = 1.602.343.8459 # Cell = 1.602.323.7608 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
tzafrir,
thanks for the note... yep, it is useless...
daveC
Tzafrir Cohen wrote:
On Tue, Jan 01, 2008 at 11:27:54AM -0500, dave cantera wrote:
vincent,
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in the current
directory...
creates a logfile, although trivial...
daveC
#!/bin/sh
#
#convert-all.sh
#
#convert all *.wav files to .gsm .au formats
#
if [ null${1} == null ]
then
FILE_LIST=`ls *.wav`
else
FILE_LIST=`ls ${1}*.wav`
fi
LOG=./log_convert.log
echo === ${LOG}
echo started at `date` ${LOG}
echo Removing all current .gsm files...
rm -f *.gsm
# A note from the Useless Use of ls Committee:
for FNAME in $1*.wav
for FNAME in ${FILE_LIST}
do
echo --- -
echo ${LOG}
echo Processing ${FNAME}...
echo Processing ${FNAME}... ${LOG}
BASEFNAME=`echo ${FNAME} | awk '{print substr($0,1,length($0)-4)}'`
echo making ${BASEFNAME}.gsm...
echo making ${BASEFNAME}.gsm... ${LOG}
#sox -q -V -c 1 ${FNAME} -r 8000 -c 1 -w ${BASEFNAME}.gsm resample -ql
2${LOG}
sox -q -V ${FNAME} -r 8000 -c 1 ${BASEFNAME}.gsm resample -ql 2${LOG}
echo ${LOG}
echo making ${BASEFNAME}.au...
echo making ${BASEFNAME}.au... ${LOG}
sox -q -V ${FNAME} -t au -r 8000 -c 1 -w ${BASEFNAME}.au resample -ql
2$
{LOG}
done
--
My wife's sister is in California.
I should buy her a Videophone2008!
Truly, The Next Best Thing to Being There!
--
WorldWideVideoPhones.com
856.380.0894
___
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[asterisk-users] Password protect a queue from callers?
Hi, We currently testing a trixbox/asterisk installation and have used Freepbx to set-up and configure the box and it is running tremendously well. We have an generic IVR configured to which can transfer callers to a child IVR. This child IVR has a number of options to send the caller off to various queues. However we would like to protect some of the options with a password/pin number so that only callers with a valid pin can gain access to the queues. We have looked at pin sets but this doesn't seem to be the correct route, we also searched on the Internet but the queries we have used bring up options to password protect queues from agents not callers. Initially we are happy with using just the one pin number for all customers for testing purposes then to proceed to using a sql backend to allow us to uniquely give each customer their own pin. Has anyone any suggestions on which direction we should proceed; we presume it could be the customer_app route which from our limited experience will hook into a custom asterisk script but being new to freepbx and asterisk we are not sure how to go about implementing this solution or if it has already been done? Any advice is appreciated... _ Fancy some celeb spotting? https://www.celebmashup.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Gotoiftime help
my final ivr is this, he works me very well exten = 110,1,GotoIfTime(08:00-18:00|mon-fri|*|*?110,in) exten =110,n,Dial(SIP/111,86,Tt) exten =110,n,Dial(SIP/112,86,Tt) exten =110,n,Hangup() exten = 110,n(in),Set(TIMEOUT(digit)=2) exten = 110,1,Answer() exten = 110,2,Background(introm) exten = 110,3,Dial(SIP/111,16,Tt) exten = 110,4,hangup thank , good example ... greetingsss -- Forwarded message -- From: Doug Lytle [EMAIL PROTECTED] I've just tested a simple include. It worked fine. My simple test: [sip-utilities] include = test1|16:50-16:55|mon-sat|*|* include = test2|16:56-16:59|mon-sat|*|* [test1] exten = 15,1,Dial(IAX2/asterisk.cw:[EMAIL PROTECTED]/5700,,t) [test2] exten = 15,1,Dial(IAX2/asterisk.cw:[EMAIL PROTECTED]/4180,,t) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Password protect a queue from callers?
I put something together like this for a finance company - Asterisk looked up the callerid in a MySQL database, and put the call into a queue, with a higher priority if the call was from certain clients. If the callerid was not found, it them allowed for a clientid and pincode to be entered. The only thing someone else wrote was the web front end to manage it all. PaulH On Wed, 2008-01-02 at 02:06 +, Caza Henha wrote: Hi, We currently testing a trixbox/asterisk installation and have used Freepbx to set-up and configure the box and it is running tremendously well. We have an generic IVR configured to which can transfer callers to a child IVR. This child IVR has a number of options to send the caller off to various queues. However we would like to protect some of the options with a password/pin number so that only callers with a valid pin can gain access to the queues. We have looked at pin sets but this doesn't seem to be the correct route, we also searched on the Internet but the queries we have used bring up options to password protect queues from agents not callers. Initially we are happy with using just the one pin number for all customers for testing purposes then to proceed to using a sql backend to allow us to uniquely give each customer their own pin. Has anyone any suggestions on which direction we should proceed; we presume it could be the customer_app route which from our limited experience will hook into a custom asterisk script but being new to freepbx and asterisk we are not sure how to go about implementing this solution or if it has already been done? Any advice is appreciated... _ Fancy some celeb spotting? https://www.celebmashup.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Gotoiftime help
On Tuesday 01 January 2008 20:40:19 troxlinux wrote: my final ivr is this, he works me very well exten = 110,1,GotoIfTime(08:00-18:00|mon-fri|*|*?110,in) exten =110,n,Dial(SIP/111,86,Tt) exten =110,n,Dial(SIP/112,86,Tt) exten =110,n,Hangup() exten = 110,n(in),Set(TIMEOUT(digit)=2) Uh, everything after this point is inoperable, given that you're reusing priority numbers. If you wanted these to actually work, you need to change the 1,2,3,4 to n,n,n,n. Or use a different extension. exten = 110,1,Answer() exten = 110,2,Background(introm) exten = 110,3,Dial(SIP/111,16,Tt) exten = 110,4,hangup -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
On Tue, 01 Jan 2008 16:10:47 +0100, MatsK [EMAIL PROTECTED] wrote: The codec is specified (for a sip device) in sip.conf, like this: Good to know. Actually, I'll have Asterisk save voicemails as WAV and move the files to the www's htdocs, and send an e-mail to users with the link they'll just have to click to listen to them. Actually, I'm thinking of embedding a Flash player in the web page, and update its playlist file so that the browser doesn't launch the external app that is registered with Windows to play WAV files. But I haven't found any Flash player that can play WAV, only MP3 :-/ And you know that you can convert the files to every codec format that is in use then will the cpu load be minimalized ! Yup, but the CPU is just a Pentium 233MHz. I just converted a 20MB WAV file from a CD-quality (44KHz sample rate, stereo) into the format Asterisk likes (8HKz, mono), and it took about 10mn. So conversion is out of the question, as Asterisk is likely to have a problem answering other incoming calls while it's busy converting the last voicemail message. To convert between different codec formats can you use the asterisk CLI command: Thanks, I didn't know this command. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
On Tue, 01 Jan 2008 11:27:54 -0500, dave cantera [EMAIL PROTECTED] wrote: here is a script that I used to convert a single wav file or the entire directory... no file specified on launch, converts all files in the current directory... Thanks for the script. I'll keep it handy. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?
On Tue, 1 Jan 2008 21:05:11 +0530, Godson Gera [EMAIL PROTECTED] wrote: Asterisk automatically takes care of saving CPU issue as it picks the file that have less translation cost Yes, but that's OK for files that I use in the IVR, but not for voicemail messages. The CPU is too slow to handle WAV-to-something_more_compact conversion in a timely manner. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
I think perhaps you are the exception rather than the rule. Maybe you were able to engineer your network so that fax works without any of the FoIP protocols - good luck to you if you have. For /most/ people, it's unlikely they would have sufficient control over their WAN segment to ensure that it is sufficiently fast and reliable enough for fax to work reliably. In any case, why on earth would you attempt to re-invent the wheel? T.38 is not only considerably more reliable and robust, it's nowhere /near/ as bandwidth intensive as G711. The original question was regarding using IAXmodem and Hylafax to receive faxes over a SIP connection. Given that T.38 can not work in this situation, the simple answer is that it /isn't/ the best solution. Jonn R Taylor wrote: NOT true and I have proven that for the last year. Jonn *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Rob Hillis *Sent:* Tuesday, January 01, 2008 4:13 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk 1.4 Fax I'd say consider yourself very lucky. I know I did some testing here some time ago with faxing over VoIP. * One extension to another over G711a with both extensions on the same LAN - worked 95% of the time * One extension on my Asterisk server to an Extension on a friend's Asterisk server using G711a via IAX - 95% failure rate. Both of us awere on the same ISP and had ping times of ~40ms between us. However, in a live environment, I convert a PSTN call to a t.38 encoded call and can send the fax just about anywhere I damn well want (where the remote end supports t.38) with a 95% success rate. t.38 is the key to successful faxing over a VoIP network. Without it, you're begging for trouble. Doug Lytle wrote: Jonn R Taylor wrote: If I had ANY failed faxes I would here about it. Iaxmodem creates a log of its own, so when I get a connection that fails hylafax sends the failure to me. One of the things that I found is you need to add nojitterbuffer to the iaxmodem config file, Really? I'll have to do some testing, I've never tried since I've read you can't. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Is Cisco 7960 SIP firmware same as 7940 SIP firmware?
Mike Dent wrote: Hi, just wondered if it was the same firmware on both devices? thanks Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yes ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trixbox and mail2fax
Hi there, is there any howto how do i configure a asterisk/trixbox for mail2fax? The fax must be send over sipgate or other SIP peers. (i dont have any normal telephones connected). What i wanne do is somethink like this: Subject: +49691234567 Attache: *.pdf The attched pdf have to be send ;) -- Mit freundlichen Grüßen Daniel mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox and mail2fax
On Jan 2, 2008 12:23 AM, Daniel [EMAIL PROTECTED] wrote: Hi there, is there any howto how do i configure a asterisk/trixbox for mail2fax? The fax must be send over sipgate or other SIP peers. (i dont have any normal telephones connected). Do people even read the mail list anymore, or do they just land on this planet, subscribe to the list, and ask the same questions that's been asked over and over and over and over and over and over Read the archives, then ask questions! Or, at the minimum, take a look at a conversation that's been going on over the past two days or so. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox and mail2fax
2008/1/2, Daniel [EMAIL PROTECTED]: Hi there, is there any howto how do i configure a asterisk/trixbox for mail2fax? The fax must be send over sipgate or other SIP peers. (i dont have any normal telephones connected). What i wanne do is somethink like this: Subject: +49691234567 Attache: *.pdf The attched pdf have to be send ;) -- Mit freundlichen Grüßen Daniel mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://jonnt.users.taylortelephone.com/trixbox/trixbox-hylafax-setup.htm. As I know , AvantFax is in trixbox... So, use a google, or write your own scripts, as I did -- Pagarbiai / Best Regards, Giedrius Augys ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk
Hi List; I heared that IAX is good for NATing issues, but I do not know if it can help me in that senario: I have two Asterisks machines in different sites and both are behind NAT (both have private IP address), I need to link these two asterisks with IAX trunk (if it help really in such senario), but I do not know if it will work without doing special routing settings on the router (like TCP/UDP port mapping or IP forwarding)? How that will be it if possible? Or I have to do a kind of port mapping? If I will need to use port mapping, then I have to map the TCP and UDP ports that are determined in iax.conf and rtp.conf files at site A for asterisk ip address at site A? Or I have to map the TCP and UDP ports that are in iax.conf and rtp.conf at site B for asterisk ip address at site A? In other words, if I am at site B then I have to go for router B and do mapping for TCP/UDP ports of the asterisk at site B or the asterisk at site A? Any help. Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init
If you have zaptel 1.2.22.1 and kernel 2.6.22 could you please do the following and see if it does the same for you? # modprobe wctdm24xxp FATAL: Error inserting wctdm24xxp (/lib/modules/2.6.22-gentoo-r9/misc/wctdm24xxp.ko): Unknown symbol in module, or unknown parameter (see dmesg) dmesg: wctdm24xxp: Unknown symbol pci_module_init Thanks Vieri --- Vieri [EMAIL PROTECTED] wrote: Hi, Before I report a bug on http://bugs.digium.com, I would like to know if someone is seeing the same error message. Personally I am not using wctdm24xxp but other modules such as wcte12xp and wctdm. The latter modules load fine and are compiled with pci_register_driver as expected. The only module that seems to require the deprecated function pci_module_init is wctdm24xxp. Is this normal? Thanks, Vieri Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox and mail2fax
Apparently not. I'm sure as heck not going to get involved in this argument again! :) Bill Hackensack wrote: Do people even read the mail list anymore, or do they just land on this planet, subscribe to the list, and ask the same questions that's been asked over and over and over and over and over and over Read the archives, then ask questions! Or, at the minimum, take a look at a conversation that's been going on over the past two days or so. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
