Re: [asterisk-users] Faxes stopped working - AMI issue?
On 12/3/19 4:21 PM, Joshua C. Colp wrote: > You'd be getting more from AMI than just that. A message comes back when > you login, another when you actually do the originate, as well as events > most likely before then. The Asterisk testsuite uses AMI heavily, as do > others, so I'm confident that AMI itself is working in that regard. > You'd need to look at your own AMI usage and ensure you are reading from > the socket and logging out. You could even connect over telnet and do > the same AMI action for Originate and test that way. I stumbled upon a message that mentioned that the interface needs two line feeds before it acts on the commands. I added that and now I can connect. I am still having some issues related to T.38 but at least the commands are making it to the server now. Thanks for all the help. -- D'Arcy J.M. Cain Vybe Networks Inc. A unit of Excelsior Solutions Corporation - Propelling Business Forward http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes stopped working - AMI issue?
On Tue, Dec 3, 2019 at 6:09 PM D'Arcy Cain wrote: > On 12/3/19 3:04 PM, Joshua C. Colp wrote: > > > The AMI command, after the login, looks like this: > > > > > > Action: Originate > > > Channel: SIP/outgoing/%%(destination)s > > > Context: LocalSets > > > CallerID: Vybe Consulting Inc Fax Service <551212> > > > Exten: sendfax > > > Priority: 1 > > > Timeout: 3 > > > Variable: faxfile=%%(faxfile)s > > > Variable: uid=%%(uid)s > > > Variable: destination=%%(destination)s > > > Variable: sender_name=Vybe Consulting Inc Fax Service > > > Variable: sender_num=55121 > Have you tried narrowing it > down at all? Using the CLI to do a test > > "channel originate" using the same dial string? Have you looked at what > > comes back from AMI as a result of the Originate to see if it shows > > anything? > > First of all, I really appreciate the help. > > So I tried this in the CLI: (everything is on one line) > > channel originate SIP/outgoing/555666 extension=sendfax@LocalSets > faxfile=/path/to/tiff/file priority=1 > The CLI command does not allow setting variables. It was more of a general test to confirm the SIP portion and originating in general. The aim is to narrow things down and verify things piece by piece. > > But I got "SendFAX requires an argument". I tried various things but I > am poking around in the dark. Can you suggest an actual CLI command > based on those options that I can try? > > Also, is there some way to have the AMI interface send back the same > errors? All I get from the socket now is "Asterisk Call Manager/5.0.1". > You'd be getting more from AMI than just that. A message comes back when you login, another when you actually do the originate, as well as events most likely before then. The Asterisk testsuite uses AMI heavily, as do others, so I'm confident that AMI itself is working in that regard. You'd need to look at your own AMI usage and ensure you are reading from the socket and logging out. You could even connect over telnet and do the same AMI action for Originate and test that way. The "SendFAX requires an argument" is a normal log message which does not go to manager. -- Joshua C. Colp Senior Software Developer Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes stopped working - AMI issue?
On 12/3/19 3:04 PM, Joshua C. Colp wrote: > The AMI command, after the login, looks like this: > > Action: Originate > Channel: SIP/outgoing/%%(destination)s > Context: LocalSets > CallerID: Vybe Consulting Inc Fax Service <551212> > Exten: sendfax > Priority: 1 > Timeout: 3 > Variable: faxfile=%%(faxfile)s > Variable: uid=%%(uid)s > Variable: destination=%%(destination)s > Variable: sender_name=Vybe Consulting Inc Fax Service > Variable: sender_num=55121 > Have you tried narrowing it down at all? Using the CLI to do a test "channel originate" using the same dial string? Have you looked at what comes back from AMI as a result of the Originate to see if it shows anything? First of all, I really appreciate the help. So I tried this in the CLI: (everything is on one line) channel originate SIP/outgoing/555666 extension=sendfax@LocalSets faxfile=/path/to/tiff/file priority=1 But I got "SendFAX requires an argument". I tried various things but I am poking around in the dark. Can you suggest an actual CLI command based on those options that I can try? Also, is there some way to have the AMI interface send back the same errors? All I get from the socket now is "Asterisk Call Manager/5.0.1". Thanks again. -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:da...@vex.net VoIP: sip:da...@vybenetworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes stopped working - AMI issue?
On Tue, Dec 3, 2019 at 4:53 PM D'Arcy Cain wrote:
> On 12/2/19 11:52 AM, Joshua C. Colp wrote:
> > So I know that AMI is listening and I can talk to it. Here is the
> > main log"
> >
> > [Nov 27 06:16:00] VERBOSE[101155] asterisk.c: Remote UNIX connection
> > [Nov 27 06:16:00] VERBOSE[101245] asterisk.c: Remote UNIX connection
> > disconnected
> > [Nov 27 06:16:01] VERBOSE[101244] manager.c: Manager 'asterisk'
> logged
> > on from 127.0.0.1
> >
> > The AMI command, after the login, looks like this:
> >
> > Action: Originate
> > Channel: SIP/outgoing/%%(destination)s
> > Context: LocalSets
> > CallerID: Vybe Consulting Inc Fax Service <551212>
> > Exten: sendfax
> > Priority: 1
> > Timeout: 3
> > Variable: faxfile=%%(faxfile)s
> > Variable: uid=%%(uid)s
> > Variable: destination=%%(destination)s
> > Variable: sender_name=Vybe Consulting Inc Fax Service
> > Variable: sender_num=551212
> >
> > Those "%%" strings get replaced by real data. My sendfax extension
> has
> > a bunch of stuff but the very first line is this:
> >
> > exten => sendfax,1,Verbose(0,FAX ${faxfile} to ${destination})
> >
> > So, regardless of what follows, shouldn't I be seeing that message
> in my
> > logs?
> >
> > Only if it was actually answered. You'd need to dig deeper by looking at
> > the SIP trace itself (sip set debug on) to see if an attempt was made
> > and what occurred.
>
> OK, so I did that and my logs are really busy now. However, a search
> through them doesn't find the phone number that I am faxing to.
>
> Are you sure that it needs to be answered before it starts logging?
> Logging only when something works isn't much of a debugging tool.
>
The way an Originate works is that upon answer the channel is directed to
the dialplan at the given target. Since your dialplan has a verbose
message, the verbose message would only be executed upon dialplan execution
which would then only occur as a result of being answered.
Have you tried narrowing it down at all? Using the CLI to do a test
"channel originate" using the same dial string? Have you looked at what
comes back from AMI as a result of the Originate to see if it shows
anything?
--
Joshua C. Colp
Senior Software Developer
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
--
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Re: [asterisk-users] Faxes stopped working - AMI issue?
On 12/2/19 11:52 AM, Joshua C. Colp wrote:
So I know that AMI is listening and I can talk to it. Here is the
main log"
[Nov 27 06:16:00] VERBOSE[101155] asterisk.c: Remote UNIX connection
[Nov 27 06:16:00] VERBOSE[101245] asterisk.c: Remote UNIX connection
disconnected
[Nov 27 06:16:01] VERBOSE[101244] manager.c: Manager 'asterisk' logged
on from 127.0.0.1
The AMI command, after the login, looks like this:
Action: Originate
Channel: SIP/outgoing/%%(destination)s
Context: LocalSets
CallerID: Vybe Consulting Inc Fax Service <551212>
Exten: sendfax
Priority: 1
Timeout: 3
Variable: faxfile=%%(faxfile)s
Variable: uid=%%(uid)s
Variable: destination=%%(destination)s
Variable: sender_name=Vybe Consulting Inc Fax Service
Variable: sender_num=551212
Those "%%" strings get replaced by real data. My sendfax extension has
a bunch of stuff but the very first line is this:
exten => sendfax,1,Verbose(0,FAX ${faxfile} to ${destination})
So, regardless of what follows, shouldn't I be seeing that message in my
logs?
Only if it was actually answered. You'd need to dig deeper by looking at
the SIP trace itself (sip set debug on) to see if an attempt was made
and what occurred.
OK, so I did that and my logs are really busy now. However, a search
through them doesn't find the phone number that I am faxing to.
Are you sure that it needs to be answered before it starts logging?
Logging only when something works isn't much of a debugging tool.
--
D'Arcy J.M. Cain
Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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Re: [asterisk-users] Faxes stopped working - AMI issue?
On Wed, Nov 27, 2019 at 3:31 PM D'Arcy Cain wrote:
> I recently upgraded from Asterisk 13.19 to 16.6.1. Everything is
> working fine with a few minor tweaks except outgoinf fax. Incoming
> works fine.
>
> I do outgoing faxing through an AMI call. Here is the output from the
> security log:
>
> [Nov 27 06:16:05] SECURITY[101222] res_security_log.c:
>
> SecurityEvent="ChallengeSent",EventTV="2019-11-27T06:16:05.566-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="alex",SessionID="0x80ba54820",LocalAddress="IPV4/UDP/
> 98.158.139.74/5060",RemoteAddress="IPV4/UDP/72.143.94.110/5060
> ",Challenge="215351b4"
> [Nov 27 06:16:05] SECURITY[101222] res_security_log.c:
>
> SecurityEvent="SuccessfulAuth",EventTV="2019-11-27T06:16:05.591-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="alex",SessionID="0x80ba54820",LocalAddress="IPV4/UDP/
> 98.158.139.74/5060",RemoteAddress="IPV4/UDP/72.143.94.110/5060
> ",UsingPassword="1"
>
> So I know that AMI is listening and I can talk to it. Here is the main
> log"
>
> [Nov 27 06:16:00] VERBOSE[101155] asterisk.c: Remote UNIX connection
> [Nov 27 06:16:00] VERBOSE[101245] asterisk.c: Remote UNIX connection
> disconnected
> [Nov 27 06:16:01] VERBOSE[101244] manager.c: Manager 'asterisk' logged
> on from 127.0.0.1
>
> The AMI command, after the login, looks like this:
>
> Action: Originate
> Channel: SIP/outgoing/%%(destination)s
> Context: LocalSets
> CallerID: Vybe Consulting Inc Fax Service <551212>
> Exten: sendfax
> Priority: 1
> Timeout: 3
> Variable: faxfile=%%(faxfile)s
> Variable: uid=%%(uid)s
> Variable: destination=%%(destination)s
> Variable: sender_name=Vybe Consulting Inc Fax Service
> Variable: sender_num=551212
>
> Those "%%" strings get replaced by real data. My sendfax extension has
> a bunch of stuff but the very first line is this:
>
> exten => sendfax,1,Verbose(0,FAX ${faxfile} to ${destination})
>
> So, regardless of what follows, shouldn't I be seeing that message in my
> logs?
>
Only if it was actually answered. You'd need to dig deeper by looking at
the SIP trace itself (sip set debug on) to see if an attempt was made and
what occurred.
--
Joshua C. Colp
Senior Software Developer
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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[asterisk-users] Faxes stopped working - AMI issue?
I recently upgraded from Asterisk 13.19 to 16.6.1. Everything is
working fine with a few minor tweaks except outgoinf fax. Incoming
works fine.
I do outgoing faxing through an AMI call. Here is the output from the
security log:
[Nov 27 06:16:05] SECURITY[101222] res_security_log.c:
SecurityEvent="ChallengeSent",EventTV="2019-11-27T06:16:05.566-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="alex",SessionID="0x80ba54820",LocalAddress="IPV4/UDP/98.158.139.74/5060",RemoteAddress="IPV4/UDP/72.143.94.110/5060",Challenge="215351b4"
[Nov 27 06:16:05] SECURITY[101222] res_security_log.c:
SecurityEvent="SuccessfulAuth",EventTV="2019-11-27T06:16:05.591-0500",Severity="Informational",Service="SIP",EventVersion="1",AccountID="alex",SessionID="0x80ba54820",LocalAddress="IPV4/UDP/98.158.139.74/5060",RemoteAddress="IPV4/UDP/72.143.94.110/5060",UsingPassword="1"
So I know that AMI is listening and I can talk to it. Here is the main log"
[Nov 27 06:16:00] VERBOSE[101155] asterisk.c: Remote UNIX connection
[Nov 27 06:16:00] VERBOSE[101245] asterisk.c: Remote UNIX connection
disconnected
[Nov 27 06:16:01] VERBOSE[101244] manager.c: Manager 'asterisk' logged
on from 127.0.0.1
The AMI command, after the login, looks like this:
Action: Originate
Channel: SIP/outgoing/%%(destination)s
Context: LocalSets
CallerID: Vybe Consulting Inc Fax Service <551212>
Exten: sendfax
Priority: 1
Timeout: 3
Variable: faxfile=%%(faxfile)s
Variable: uid=%%(uid)s
Variable: destination=%%(destination)s
Variable: sender_name=Vybe Consulting Inc Fax Service
Variable: sender_num=551212
Those "%%" strings get replaced by real data. My sendfax extension has
a bunch of stuff but the very first line is this:
exten => sendfax,1,Verbose(0,FAX ${faxfile} to ${destination})
So, regardless of what follows, shouldn't I be seeing that message in my
logs?
--
D'Arcy J.M. Cain
Vybe Networks Inc.
A unit of Excelsior Solutions Corporation - Propelling Business Forward
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Faxes suddenly failing
You should try disabling ecm.. put the following in res_fax.conf ; Enable/disable T.30 ECM (error correction mode) by default. ; Default: Enabled ecm=no On Thu, Sep 1, 2011 at 1:27 AM, C F shma...@gmail.com wrote: I think you should change the subject line to: Faxes suddenly worked for 2 weeks. On Wed, Aug 31, 2011 at 3:49 PM, Tim King tim.compnetw...@gmail.com wrote: I realize that faxing is not great with voip but here is my confusion. I have been working on a web based fax system for 2 weeks. During this time I have sent over 100 2 page faxes without any errors. Now today as things are finally completed I can not seem to get any fax to go through unless it is a 1 page cover only. Anyone able to tell the issue from this debug output? -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX st: IDLE rt: IDLENSRX -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY rt: RRDYNHRY -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.091837 ], stack sent 5 frames (100 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.160248 ], stack sent 3 frames (60 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.960201 ], channel sent 48 frames (960 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.979464 ], channel sent 1 frames (20 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.157848 ], stack sent 150 frames (3000 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.219814 ], stack sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 005.579811 ], stack sent 118 frames (2360 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 006.481179 ], channel sent 275 frames (5500 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 007.801045 ], channel sent 66 frames (1320 ms) of energy. -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP st: WT_DIS_RSP rt: NT4X -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP rt: UNEXPECT -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_ENDst: WT_DIS_RSP rt: RXXXNFRX -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 011.152812 ], stack sent 279 frames (5580 ms) of silence. -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 013.471827 ], stack sent 116 frames (2320 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 014.260642 ], channel sent 323 frames (6460 ms) of silence. -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.460661 ], channel sent 110 frames (2200 ms) of energy. -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS st: WT_DIS_RSP rt: WDSRNDCS -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400 -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4 -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196 -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECMst: WT_DIS_RSP rt: WDSRNSWE Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.540315 ], channel sent 4 frames (80 ms) of silence. -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_TRN rt: UNEXPECT Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.700543 ], channel sent 158 frames (3160 ms) of energy. -- FAX handle 0: [ 019.759984 ], STAT_EVT_RX_TRN_ENDst: RCV_ECM_TRN rt: RTCFNERT -- FAX handle 0: [ 019.760071 ], STAT_FRM_CFR Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.912812 ], stack sent 322 frames (6440 ms) of silence. -- FAX handle 0: [ 020.957834 ], STAT_EVT_TX_V21_DONE st: RCV_ECM_STRT rt: RECMNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 021.278809 ], stack sent 68 frames (1360 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 022.261160 ], channel sent 128 frames (2560 ms) of
Re: [asterisk-users] Faxes suddenly failing
That debug looks cool but I have no idea what it means. If you are using T.38, turn it off, and do audio fax, recorded with MixMonitor. When you can hear the audio of the fax hopefully you will be able to tell what's going on, and if you're lucky it's something specific to the particular kind of testing you are doing. I don't think it's an exaggeration to say there have been hundreds of posts over the last few years about broken T.38. Avoid it in favor of traditional audio faxing. Even if you can control both endpoints, there's just so much that can go wrong when faxing over voip. If you need this to be 'reliable faxing', you should seriously consider doing your faxes over copper. If you cannot afford that, it should be a top-tier voip provider on a dedicated line, where you will not be starving for bandwidth, and you should never compress the audio on those calls. On Wed, Aug 31, 2011 at 7:27 PM, C F shma...@gmail.com wrote: I think you should change the subject line to: Faxes suddenly worked for 2 weeks. On Wed, Aug 31, 2011 at 3:49 PM, Tim King tim.compnetw...@gmail.com wrote: I realize that faxing is not great with voip but here is my confusion. I have been working on a web based fax system for 2 weeks. During this time I have sent over 100 2 page faxes without any errors. Now today as things are finally completed I can not seem to get any fax to go through unless it is a 1 page cover only. Anyone able to tell the issue from this debug output? -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX st: IDLE rt: IDLENSRX -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY rt: RRDYNHRY -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.091837 ], stack sent 5 frames (100 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.160248 ], stack sent 3 frames (60 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.960201 ], channel sent 48 frames (960 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.979464 ], channel sent 1 frames (20 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.157848 ], stack sent 150 frames (3000 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.219814 ], stack sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 005.579811 ], stack sent 118 frames (2360 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 006.481179 ], channel sent 275 frames (5500 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 007.801045 ], channel sent 66 frames (1320 ms) of energy. -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP st: WT_DIS_RSP rt: NT4X -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP rt: UNEXPECT -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_END st: WT_DIS_RSP rt: RXXXNFRX -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 011.152812 ], stack sent 279 frames (5580 ms) of silence. -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 013.471827 ], stack sent 116 frames (2320 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 014.260642 ], channel sent 323 frames (6460 ms) of silence. -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.460661 ], channel sent 110 frames (2200 ms) of energy. -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS st: WT_DIS_RSP rt: WDSRNDCS -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400 -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4 -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196 -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECM st: WT_DIS_RSP rt: WDSRNSWE Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.540315 ], channel sent 4 frames (80 ms) of silence. -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_TRN rt: UNEXPECT Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.700543 ], channel sent 158 frames
Re: [asterisk-users] Faxes suddenly failing
Hi Tim, On 09/01/2011 03:49 AM, Tim King wrote: I realize that faxing is not great with voip but here is my confusion. I have been working on a web based fax system for 2 weeks. During this time I have sent over 100 2 page faxes without any errors. Now today as things are finally completed I can not seem to get any fax to go through unless it is a 1 page cover only. Anyone able to tell the issue from this debug output? -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX st: IDLE rt: IDLENSRX -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY rt: RRDYNHRY -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.091837 ], stack sent 5 frames (100 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.160248 ], stack sent 3 frames (60 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.960201 ], channel sent 48 frames (960 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.979464 ], channel sent 1 frames (20 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.157848 ], stack sent 150 frames (3000 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.219814 ], stack sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 005.579811 ], stack sent 118 frames (2360 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 006.481179 ], channel sent 275 frames (5500 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 007.801045 ], channel sent 66 frames (1320 ms) of energy. -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP st: WT_DIS_RSP rt: NT4X -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP rt: UNEXPECT -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_ENDst: WT_DIS_RSP rt: RXXXNFRX -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 011.152812 ], stack sent 279 frames (5580 ms) of silence. -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 013.471827 ], stack sent 116 frames (2320 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 014.260642 ], channel sent 323 frames (6460 ms) of silence. -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.460661 ], channel sent 110 frames (2200 ms) of energy. -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS st: WT_DIS_RSP rt: WDSRNDCS -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400 -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4 -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196 -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECMst: WT_DIS_RSP rt: WDSRNSWE Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.540315 ], channel sent 4 frames (80 ms) of silence. -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_TRN rt: UNEXPECT Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.700543 ], channel sent 158 frames (3160 ms) of energy. -- FAX handle 0: [ 019.759984 ], STAT_EVT_RX_TRN_ENDst: RCV_ECM_TRN rt: RTCFNERT -- FAX handle 0: [ 019.760071 ], STAT_FRM_CFR Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.912812 ], stack sent 322 frames (6440 ms) of silence. -- FAX handle 0: [ 020.957834 ], STAT_EVT_TX_V21_DONE st: RCV_ECM_STRT rt: RECMNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 021.278809 ], stack sent 68 frames (1360 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 022.261160 ], channel sent 128 frames (2560 ms) of silence. -- FAX handle 0: [ 022.517880 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_STRT rt: RECMNSRI -- FAX handle 0: [ 022.517982 ], P30EVN_PHASE_C -- FAX handle 0: [ 022.517998 ], P30EVN_DOC_START -- FAX handle 0: [ 022.518429 ], P30EVN_PAGE_START Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 031.102000 ], channel sent 442 frames (8840 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 031.160415 ], channel sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 031.160196 ], STAT_EVT_RX_IMG_ENDst: RCV_ECM
Re: [asterisk-users] Faxes suddenly failing
kirsten du toit wrote: You should try disabling ecm.. This seems crazy to me. Why are you recommending it? Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes suddenly failing
On 09/01/2011 11:50 PM, Lee Howard wrote: kirsten du toit wrote: You should try disabling ecm.. This seems crazy to me. Why are you recommending it? Because its the industry standard last resort of anyone who doesn't understand FAX and is using T.38. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes suddenly failing
Steve Underwood wrote: On 09/01/2011 11:50 PM, Lee Howard wrote: kirsten du toit wrote: You should try disabling ecm.. This seems crazy to me. Why are you recommending it? Because its the industry standard last resort of anyone who doesn't understand FAX and is using T.38. Even HP recommends for their own fax machines it numerous times: http://h2.www2.hp.com/bizsupport/TechSupport/Document.jsp?lang=encc=ustaskId=110prodSeriesId=378056prodTypeId=18972objectID=c00062808 http://h2.www2.hp.com/bizsupport/TechSupport/Document.jsp?objectID=buu02549lang=encc=uscontentType=SupportFAQprodSeriesId=3366988prodTypeId=15179 Yes, always a last-ditch effort, and if it actually succeeds in getting a legible document through then it means that either 1) the ECM protocol on either the sender or the receiver is gravely flawed, or 2) something that requires ECM (like V.34-Fax/SuperG3) ended up being disabled along with ECM and that the problem really had to do with that something and not with ECM. I've never seen a fax document that couldn't make it through with ECM enabled be able to come through legibly with ECM disabled otherwise. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Faxes suddenly failing
I realize that faxing is not great with voip but here is my confusion. I have been working on a web based fax system for 2 weeks. During this time I have sent over 100 2 page faxes without any errors. Now today as things are finally completed I can not seem to get any fax to go through unless it is a 1 page cover only. Anyone able to tell the issue from this debug output? -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX st: IDLE rt: IDLENSRX -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY rt: RRDYNHRY -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.091837 ], stack sent 5 frames (100 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.160248 ], stack sent 3 frames (60 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.960201 ], channel sent 48 frames (960 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.979464 ], channel sent 1 frames (20 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.157848 ], stack sent 150 frames (3000 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.219814 ], stack sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 005.579811 ], stack sent 118 frames (2360 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 006.481179 ], channel sent 275 frames (5500 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 007.801045 ], channel sent 66 frames (1320 ms) of energy. -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP st: WT_DIS_RSP rt: NT4X -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP rt: UNEXPECT -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_ENDst: WT_DIS_RSP rt: RXXXNFRX -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 011.152812 ], stack sent 279 frames (5580 ms) of silence. -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 013.471827 ], stack sent 116 frames (2320 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 014.260642 ], channel sent 323 frames (6460 ms) of silence. -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.460661 ], channel sent 110 frames (2200 ms) of energy. -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS st: WT_DIS_RSP rt: WDSRNDCS -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400 -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4 -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196 -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECMst: WT_DIS_RSP rt: WDSRNSWE Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.540315 ], channel sent 4 frames (80 ms) of silence. -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_TRN rt: UNEXPECT Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.700543 ], channel sent 158 frames (3160 ms) of energy. -- FAX handle 0: [ 019.759984 ], STAT_EVT_RX_TRN_ENDst: RCV_ECM_TRN rt: RTCFNERT -- FAX handle 0: [ 019.760071 ], STAT_FRM_CFR Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.912812 ], stack sent 322 frames (6440 ms) of silence. -- FAX handle 0: [ 020.957834 ], STAT_EVT_TX_V21_DONE st: RCV_ECM_STRT rt: RECMNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 021.278809 ], stack sent 68 frames (1360 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 022.261160 ], channel sent 128 frames (2560 ms) of silence. -- FAX handle 0: [ 022.517880 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_STRT rt: RECMNSRI -- FAX handle 0: [ 022.517982 ], P30EVN_PHASE_C -- FAX handle 0: [ 022.517998 ], P30EVN_DOC_START -- FAX handle 0: [ 022.518429 ], P30EVN_PAGE_START Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 031.102000 ], channel sent 442 frames (8840 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 031.160415 ], channel sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 031.160196 ],
Re: [asterisk-users] Faxes suddenly failing
I think you should change the subject line to: Faxes suddenly worked for 2 weeks. On Wed, Aug 31, 2011 at 3:49 PM, Tim King tim.compnetw...@gmail.com wrote: I realize that faxing is not great with voip but here is my confusion. I have been working on a web based fax system for 2 weeks. During this time I have sent over 100 2 page faxes without any errors. Now today as things are finally completed I can not seem to get any fax to go through unless it is a 1 page cover only. Anyone able to tell the issue from this debug output? -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX st: IDLE rt: IDLENSRX -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY rt: RRDYNHRY -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.091837 ], stack sent 5 frames (100 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.160248 ], stack sent 3 frames (60 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.960201 ], channel sent 48 frames (960 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.979464 ], channel sent 1 frames (20 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.157848 ], stack sent 150 frames (3000 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.219814 ], stack sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 005.579811 ], stack sent 118 frames (2360 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 006.481179 ], channel sent 275 frames (5500 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 007.801045 ], channel sent 66 frames (1320 ms) of energy. -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP st: WT_DIS_RSP rt: NT4X -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP rt: UNEXPECT -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_END st: WT_DIS_RSP rt: RXXXNFRX -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 011.152812 ], stack sent 279 frames (5580 ms) of silence. -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 013.471827 ], stack sent 116 frames (2320 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 014.260642 ], channel sent 323 frames (6460 ms) of silence. -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.460661 ], channel sent 110 frames (2200 ms) of energy. -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS st: WT_DIS_RSP rt: WDSRNDCS -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400 -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4 -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196 -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECM st: WT_DIS_RSP rt: WDSRNSWE Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.540315 ], channel sent 4 frames (80 ms) of silence. -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_TRN rt: UNEXPECT Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.700543 ], channel sent 158 frames (3160 ms) of energy. -- FAX handle 0: [ 019.759984 ], STAT_EVT_RX_TRN_END st: RCV_ECM_TRN rt: RTCFNERT -- FAX handle 0: [ 019.760071 ], STAT_FRM_CFR Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.912812 ], stack sent 322 frames (6440 ms) of silence. -- FAX handle 0: [ 020.957834 ], STAT_EVT_TX_V21_DONE st: RCV_ECM_STRT rt: RECMNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 021.278809 ], stack sent 68 frames (1360 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 022.261160 ], channel sent 128 frames (2560 ms) of silence. -- FAX handle 0: [ 022.517880 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_STRT rt: RECMNSRI -- FAX handle 0: [ 022.517982 ], P30EVN_PHASE_C -- FAX handle 0: [ 022.517998 ], P30EVN_DOC_START -- FAX handle 0: [ 022.518429 ], P30EVN_PAGE_START Channel 'SIP/MyVoipProvider-0046' fax
[asterisk-users] Faxes
We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI) cards. I am having trouble completing faxes. Carrier send calls to me using SIP. Any recommendation to have some success with Fax. We trying using T.38 pass through and using G711U codec. Asterisk Version 1.6.1.1 Thanks, Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
On Fri, Sep 3, 2010 at 10:49 AM, dave george dgeo...@teletoneinc.com wrote: We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI) cards. I am having trouble completing faxes. Carrier send calls to me using SIP. Any recommendation to have some success with Fax. We trying using T.38 pass through and using G711U codec. Asterisk Version 1.6.1.1 Thanks, Dave Dave, T.38 in some fashion. But you don't really explain your call flow or what you are trying to do. You say you have PSTN and then talk about SIP. Are you just trying to pass the calls to physical FAX machines, or a server to handle faxing? Elaborate a bit and I am sure someone can offer some advice. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, September 03, 2010 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faxes On Fri, Sep 3, 2010 at 10:49 AM, dave george dgeo...@teletoneinc.com wrote: We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI) cards. I am having trouble completing faxes. Carrier send calls to me using SIP. Any recommendation to have some success with Fax. We trying using T.38 pass through and using G711U codec. Asterisk Version 1.6.1.1 Thanks, Dave Dave, T.38 in some fashion. But you don't really explain your call flow or what you are trying to do. You say you have PSTN and then talk about SIP. Are you just trying to pass the calls to physical FAX machines, or a server to handle faxing? Elaborate a bit and I am sure someone can offer some advice. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
On Fri, Sep 3, 2010 at 11:50 AM, dave george dgeo...@teletoneinc.com wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. You don't say the percentage that are failing. However, people who have worked with SIP on asterisk have been known to do: exten = s,1,Playback(silence/1) exten = s,n,Whatever(is_next) And I don't know why, but this seems to make things better. If you're doing an Answer and then a receive_Fax, try putting a playback silence in between and see if that helps anything. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
All my attempts are failing. Thanks Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Friday, September 03, 2010 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Faxes On Fri, Sep 3, 2010 at 11:50 AM, dave george dgeo...@teletoneinc.com wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. You don't say the percentage that are failing. However, people who have worked with SIP on asterisk have been known to do: exten = s,1,Playback(silence/1) exten = s,n,Whatever(is_next) And I don't know why, but this seems to make things better. If you're doing an Answer and then a receive_Fax, try putting a playback silence in between and see if that helps anything. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
Can you post the dialplan snippet you are using? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
On 09/03/2010 10:50 AM, dave george wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, because the PSTN does not speak T.38. If one side of the call is SIP, and the other side is TDM, then you have only two choices: pass the call through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX over T.38). At this time, the only option without patching Asterisk is to pass the call through in audio mode, but there are many, many problems with doing FAX over VoIP (Steve Underwood's page on the soft-switch.org site explains them very well). There are patches in the issue tracker at issues.asterisk.org to add T.38 gateway functionality to various releases of Asterisk, and they work well for quite a few people. If you added that, you'd be able to act as a T.38 gateway, which would dramatically increase your chances of success. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
Thanks Kevin, I tried passing it over VOIP using g711U codecs with no success. I will try using the patches that you mentioned and post the results. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, September 03, 2010 2:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Faxes On 09/03/2010 10:50 AM, dave george wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, because the PSTN does not speak T.38. If one side of the call is SIP, and the other side is TDM, then you have only two choices: pass the call through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX over T.38). At this time, the only option without patching Asterisk is to pass the call through in audio mode, but there are many, many problems with doing FAX over VoIP (Steve Underwood's page on the soft-switch.org site explains them very well). There are patches in the issue tracker at issues.asterisk.org to add T.38 gateway functionality to various releases of Asterisk, and they work well for quite a few people. If you added that, you'd be able to act as a T.38 gateway, which would dramatically increase your chances of success. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
g711 across a network without perfect jitter/delay characteristics will not work. You cannot do g711 faxing across the internet - at all. It's not a perfect solution even in an office on a dedicated LAN environment (you'll still get failed faxes). On Fri, Sep 3, 2010 at 12:32 PM, dave george dgeo...@teletoneinc.comwrote: Thanks Kevin, I tried passing it over VOIP using g711U codecs with no success. I will try using the patches that you mentioned and post the results. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, September 03, 2010 2:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Faxes On 09/03/2010 10:50 AM, dave george wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, because the PSTN does not speak T.38. If one side of the call is SIP, and the other side is TDM, then you have only two choices: pass the call through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX over T.38). At this time, the only option without patching Asterisk is to pass the call through in audio mode, but there are many, many problems with doing FAX over VoIP (Steve Underwood's page on the soft-switch.org site explains them very well). There are patches in the issue tracker at issues.asterisk.org to add T.38 gateway functionality to various releases of Asterisk, and they work well for quite a few people. If you added that, you'd be able to act as a T.38 gateway, which would dramatically increase your chances of success. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
Try open souce solution ICTFAX for T.38 faxing developed by us available at http://www.sourceforge.net/projects/ictfax Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Sat, Sep 4, 2010 at 3:03 AM, Joel Maslak jmas...@antelope.net wrote: g711 across a network without perfect jitter/delay characteristics will not work. You cannot do g711 faxing across the internet - at all. It's not a perfect solution even in an office on a dedicated LAN environment (you'll still get failed faxes). On Fri, Sep 3, 2010 at 12:32 PM, dave george dgeo...@teletoneinc.comwrote: Thanks Kevin, I tried passing it over VOIP using g711U codecs with no success. I will try using the patches that you mentioned and post the results. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, September 03, 2010 2:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Faxes On 09/03/2010 10:50 AM, dave george wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, because the PSTN does not speak T.38. If one side of the call is SIP, and the other side is TDM, then you have only two choices: pass the call through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX over T.38). At this time, the only option without patching Asterisk is to pass the call through in audio mode, but there are many, many problems with doing FAX over VoIP (Steve Underwood's page on the soft-switch.org site explains them very well). There are patches in the issue tracker at issues.asterisk.org to add T.38 gateway functionality to various releases of Asterisk, and they work well for quite a few people. If you added that, you'd be able to act as a T.38 gateway, which would dramatically increase your chances of success. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Faxes from website works, but from regular don't: cause 16
Hi Guys, I'm having a non-obvious issue, i am using Fax for asterisk to receive faxes, so when i test using a website that send faxes it's working great: the fax is received and the fax2mail app is called and i get it in my email box. but when i try using a regular fax machine everything in logs (turned on debug) but all of the sudden a line appear saying: Channel 0/1, span 1 got hangup request, cause 16 and then the fax2mail is not called for some reason and [image: :(] no fax received, can you help me guys with that? thanks!!! -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] faxes received on mISDN
On Thu, Nov 5, 2009 at 6:27 AM, Vieri rentor...@yahoo.com wrote: Despite the simpler setup, the faxes don't come in. From the logs I can see that Asterisk receives fax calls and dials the iaxmodem (on localhost). However, no data is transmitted according to Hylafax. Modify your dialplan to record the calls. Listen to the recording. Does the call ever connect? Does it sound like garbage? When you can hear what's going wrong you should be able to make better guesses. ___ -- 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] faxes
How does a Windows workstation fax via Asterisk? Has someone written a Asterisk fax print driver? Or some other way? Angus Comber [EMAIL PROTECTED] - Original Message - From: Henry Devito [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 26, 2005 9:11 AM Subject: Re: [Asterisk-Users] faxes I've been working on, actually just started, creating a network app where windoze pc's can print to a virtual printer which in turn will make asterisk send the fax out. I also have asterisk set up for a client where all it does is send and recieve faxes. They have 14 fax machines on SPA2000 to receive faxes and then there are 40 stations connected to ATA's to send faxes out. Of course they are using multiple data T1's connected to the internet which are very stable. - Original Message - From: Michael K. Rodriguez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 25, 2005 11:29 PM Subject: Re: [Asterisk-Users] faxes I have tested a fax call on asterisk with success. I used an IAXy on a broadband Time Warner connection. Faxes are much more sensitive than voice calls. If you have a good internet connection, faxes should complete fine. The only downfall it is recommended that you call to verify fax transmission after every fax. -Michael On 3/25/05 10:59 PM, AS [EMAIL PROTECTED] wrote: Is it possible and if so for a workstation user to send his fax via asterisk? ___ 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 ___ 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 ___ 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 ___ 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] faxes
I've been working on, actually just started, creating a network app where windoze pc's can print to a virtual printer which in turn will make asterisk send the fax out. I also have asterisk set up for a client where all it does is send and recieve faxes. They have 14 fax machines on SPA2000 to receive faxes and then there are 40 stations connected to ATA's to send faxes out. Of course they are using multiple data T1's connected to the internet which are very stable. - Original Message - From: Michael K. Rodriguez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 25, 2005 11:29 PM Subject: Re: [Asterisk-Users] faxes I have tested a fax call on asterisk with success. I used an IAXy on a broadband Time Warner connection. Faxes are much more sensitive than voice calls. If you have a good internet connection, faxes should complete fine. The only downfall it is recommended that you call to verify fax transmission after every fax. -Michael On 3/25/05 10:59 PM, AS [EMAIL PROTECTED] wrote: Is it possible and if so for a workstation user to send his fax via asterisk? ___ 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 ___ 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 ___ 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] faxes
Do you know if it is possible to setup cups to have a fax printer that will use asterisk? On 26/03/2005, at 6:11 PM, Henry Devito wrote: I've been working on, actually just started, creating a network app where windoze pc's can print to a virtual printer which in turn will make asterisk send the fax out. I also have asterisk set up for a client where all it does is send and recieve faxes. They have 14 fax machines on SPA2000 to receive faxes and then there are 40 stations connected to ATA's to send faxes out. Of course they are using multiple data T1's connected to the internet which are very stable. - Original Message - From: Michael K. Rodriguez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 25, 2005 11:29 PM Subject: Re: [Asterisk-Users] faxes I have tested a fax call on asterisk with success. I used an IAXy on a broadband Time Warner connection. Faxes are much more sensitive than voice calls. If you have a good internet connection, faxes should complete fine. The only downfall it is recommended that you call to verify fax transmission after every fax. -Michael On 3/25/05 10:59 PM, AS [EMAIL PROTECTED] wrote: Is it possible and if so for a workstation user to send his fax via asterisk? ___ 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 ___ 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 ___ 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 ___ 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
[Asterisk-Users] faxes
Is it possible and if so for a workstation user to send his fax via asterisk? ___ 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] faxes
I have tested a fax call on asterisk with success. I used an IAXy on a broadband Time Warner connection. Faxes are much more sensitive than voice calls. If you have a good internet connection, faxes should complete fine. The only downfall it is recommended that you call to verify fax transmission after every fax. -Michael On 3/25/05 10:59 PM, AS [EMAIL PROTECTED] wrote: Is it possible and if so for a workstation user to send his fax via asterisk? ___ 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 ___ 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
