Re: [asterisk-users] TDM400P dialout problem
Kevin P. Fleming wrote: ... The messages in bug 12099 are *not* errors, they are annoyances only. The latest SVN branch 1.4 code of Asterisk will no longer generate them, Using today's svn 3915: .. Answer(Zap/2-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/2-1, Zap/4/2375678) in new stack -- Called 4/2375678 [Mar 2 11:15:35] WARNING[2320]: chan_zap.c:4424 zt_handle_event: Detected alarm on channel 4: Red Alarm -- Hungup 'Zap/4-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Congestion(Zap/2-1, ) in new stack [Mar 2 11:15:35] NOTICE[2320]: chan_zap.c:7514 handle_init_event: Alarm cleared on channel 4 == Spawn extension (internal, 2375678, 3) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' zap show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) sean ___ -- 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] TDM400P dialout problem
Tzafrir Cohen wrote: On Sun, Mar 02, 2008 at 11:36:03AM -0500, sean darcy wrote: Kevin P. Fleming wrote: ... The messages in bug 12099 are *not* errors, they are annoyances only. The latest SVN branch 1.4 code of Asterisk will no longer generate them, Using today's svn 3915: .. Answer(Zap/2-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/2-1, Zap/4/2375678) in new stack -- Called 4/2375678 [Mar 2 11:15:35] WARNING[2320]: chan_zap.c:4424 zt_handle_event: Detected alarm on channel 4: Red Alarm -- Hungup 'Zap/4-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:3] Congestion(Zap/2-1, ) in new stack [Mar 2 11:15:35] NOTICE[2320]: chan_zap.c:7514 handle_init_event: Alarm cleared on channel 4 == Spawn extension (internal, 2375678, 3) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' zap show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) This shows alarms on the whole span (the card). The alarms in question are alarms on specific channels. The whole interface makes more sense in the other meduims where the span usually corresponds to one physical meduim. I'm a little confused. Are you responding to my including the zap show status command ( which I did just for background), or to the call description? If you are responding to the call description, doesn't the alarm show up specifically on channel 4? In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. sean ___ -- 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] TDM400P dialout problem
Kevin P. Fleming wrote: sean darcy wrote: In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. As I've mentioned previously, the changes to fix this for good (assuming they work properly) are in http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms, although one tester has reported that incoming calls don't work properly using that branch, so it still needs some work. Sorry, I hadn't seen this mentioned. I'll try it asap. thanks for the lead. sean ___ -- 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] TDM400P dialout problem
Kevin P. Fleming wrote: sean darcy wrote: In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. As I've mentioned previously, the changes to fix this for good (assuming they work properly) are in http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms, although one tester has reported that incoming calls don't work properly using that branch, so it still needs some work. Got it. No more Red Alarms. Which is great. But...now I keep getting incomplete number messages from the co. No trouble on the console: Starting simple switch on 'Zap/2-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/2-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/2-1, Zap/4/6981000) in new stack -- Called 4/6981000 -- Zap/4-1 answered Zap/2-1 -- Native bridging Zap/2-1 and Zap/4-1 -- Hungup 'Zap/4-1' which shows the correct local number, which can be dialed from a plain telephone. It's as though the Dial command just didn't send some of the digits correctly. Thanks for all the help. sean ___ -- 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] TDM400P dialout problem
sean darcy wrote: Kevin P. Fleming wrote: sean darcy wrote: In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. As I've mentioned previously, the changes to fix this for good (assuming they work properly) are in http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms, although one tester has reported that incoming calls don't work properly using that branch, so it still needs some work. Got it. No more Red Alarms. Which is great. But...now I keep getting incomplete number messages from the co. No trouble on the console: Starting simple switch on 'Zap/2-1' -- Executing [EMAIL PROTECTED]:1] Answer(Zap/2-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(Zap/2-1, Zap/4/6981000) in new stack -- Called 4/6981000 -- Zap/4-1 answered Zap/2-1 -- Native bridging Zap/2-1 and Zap/4-1 -- Hungup 'Zap/4-1' which shows the correct local number, which can be dialed from a plain telephone. It's as though the Dial command just didn't send some of the digits correctly. And incoming calls aren't answered. No ring event. No nothing. svn-3915 incoming calls were answered, but generated a Red Alarm. sean ___ -- 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] TDM400P dialout problem
Martin wrote: . In any event, as least for me the TDM400P seems to have problems with zaptel svn - not just an annoyance. As I've mentioned previously, the changes to fix this for good (assuming they work properly) are in http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms, although How can I download this, do I need SVN installed? yes. install svn. Then: svn co http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms trunk good luck. sean ___ -- 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] ekiga sip registration fails; externip no help
ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=as64618445 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 113 REGISTER User-Agent: Asterisk PBX 1.6.0-beta4 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- --- SIP read from UDP://86.64.162.35:5060 --- SIP/2.0 406 Not Acceptable Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport=5060;received=96.xxx.253.yy From: sip:[EMAIL PROTECTED];tag=as64618445 To: sip:[EMAIL PROTECTED];tag=12d18c5009a2de32fca8ae9d70ad0321.ef55 Call-ID: [EMAIL PROTECTED] CSeq: 113 REGISTER Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 86.64.162.35:5060 Noisy feedback tells: pid=24578 req_src_ip=96.xxx.253.yy req_src_port=5060 in_uri=sip:ekiga.net out_uri=sip:ekiga.net via_cnt==1 - --- (9 headers 0 lines) --- -- Got SIP response 406 Not Acceptable back from 86.64.162.35 ___ -- 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] ekiga sip registration fails; externip no help
sean darcy wrote: ekiga registration fails. I've set nat = yes ( also blank ) and i've set externip. Anybody have a sip.conf that works? Here's the sip debug: Reliably Transmitting (NAT) to 86.64.162.35:5060: REGISTER sip:ekiga.net SIP/2.0 Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport Max-Forwards: 70 From: sip:[EMAIL PROTECTED];tag=as64618445 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 113 REGISTER User-Agent: Asterisk PBX 1.6.0-beta4 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- --- SIP read from UDP://86.64.162.35:5060 --- SIP/2.0 406 Not Acceptable Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport=5060;received=96.xxx.253.yy From: sip:[EMAIL PROTECTED];tag=as64618445 To: sip:[EMAIL PROTECTED];tag=12d18c5009a2de32fca8ae9d70ad0321.ef55 Call-ID: [EMAIL PROTECTED] CSeq: 113 REGISTER Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 86.64.162.35:5060 Noisy feedback tells: pid=24578 req_src_ip=96.xxx.253.yy req_src_port=5060 in_uri=sip:ekiga.net out_uri=sip:ekiga.net via_cnt==1 - --- (9 headers 0 lines) --- -- Got SIP response 406 Not Acceptable back from 86.64.162.35 For anyone else: 1. you need localnet = in sip.conf, e.g. localnet =10.0.0.0/255.0.0.0 2. only use ONE of externhost, externip or stunaddr. stunaddr worked for me. sean ___ -- 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] cordless usb handsets: Uniden Win1200?
I'm looking for a usb cordless handset to pair with a softphone ( probably ekiga) on a pc linked to an asterisk server. I've loooked at the bluetooth headsets, but they seem overkill for just home phone extensions. I've found a number of handsets that work with skype, but they seem locked into skype. I've also found the Uniden WIN1200, which works with MS Messenger. It doesn't appear to be locked in like as the skype phones, and doesn't Messenger use sip? Anyone have any experience or suggestions? sean ___ -- 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 not picking up (some) calls due to zaptel detecting and clearing alarms
Gonzalo Servat wrote: On Thu, Mar 27, 2008 at 1:56 PM, Tzafrir Cohen [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Any suggestions?? I'm using Asterisk 1.6.0-beta4 and Zaptel 1.4.9.2 http://1.4.9.2. A freshly-built Asterisk? Built vs. zaptel 1.4.9.2 http://1.4.9.2 ? Yes, I built 1.6.0-beta4 just recently with zaptel 1.4.9.2 http://1.4.9.2. As per your suggestion on IRC, I've checked out, compiled and installed Zaptel from SVN (1.4 branch). I reloaded the zaptel modules but ... no go. Do I need to recompile Asterisk too? Shouldn't it have picked up the alarm as a red alarm on the channel? I've no idea to be honest. (Besides the problem. Is 1.4 SVN recommended for that at the moment?) Also no idea. I was told to use 1.4 if I'm using Asterisk 1.6 so I went with that. - Gonzalo Try: /svn/zaptel/!svn/ver/3905/team/kpfleming/battery_alarms It worked for me. You should have to rebuild asterisk. We do need a new zaptel release though. sean ___ -- 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] TDM410E card, 1 FXO module - how to dial Out
Kevin P. Fleming wrote:
Mojo with Horan Company, LLC wrote:
P.S. If you can't dial seven digit numbers in your area, but you miss
it, you can restore that behavior if you feel like selecting a default
area code:
exten = _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
Here, if I dial a seven digit number, asterisk dials 907 followed by my
seven digits out the phone line.
Well, sort of. This will also trigger if you dial the first 7 digits of
a 10-digit number from a device that doesn't dial 'en bloc', since there
is no longer any way to distinguish 7-vs-10 digit numbers by the number
pattern. In other words, this will work fine if you are dialing from a
SIP phone, but not if you are dialing from an analog phone.
With some trepidation, I can say my home system doesn't seem to work
that way. Using an analog phone, I can deal 3, 7, 10 or 11 numbers and
all goes as I expect.
After seeing this post, I wondered why :). It seems * waits about 4 secs
to see if all the numbers are dialed. Or is it some fortuitous order of
the includes ( vaguely remembering posts about how extensions were
searched)?
extensions.conf:
[internal]
include = outbound-local
include = outbound-long-distance
include = office-extensions
[outbound-local]
exten = _NXX,1,Answer()
exten = _NXX,n,Dial(${faxline}/${EXTEN})
[outbound-long-distance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,Dial(iax2/office/${EXTEN})
exten =_NXXNXX,1,Answer()
exten =_NXXNXX,n,Dial(iax2/office/${EXTEN})
[office-extensions]
exten =_1XX,1,Answer()
exten =_1XX,n,Dial(iax2/office/${EXTEN})
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[asterisk-users] howto debug bad iax voice quality?
I'm set up to call 3 digit extensions at the office ( running 1.4.13) from home ( 1.6.0 beta7) over iax. 1 out of 3 times the call breaks up, but only in the home - office direction. office - home always sounds good. If it were a poor internet connection, I'd expect both sides of the conversation to be poor. Not surprisingly, each side can ping the other in the same time - 25-30ms. Both servers have the iax jitterbuffer on. I could always use a lower bit-rate codec ( now using mu-law ), but I don't see how it could be a one way bit-rate issue. Any suggestions appreciated. sean ___ -- 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] howto debug bad iax voice quality?
John Beaman wrote: John Beaman Telecom Specialist II Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 4/4/2008 3:23:49 PM I'm set up to call 3 digit extensions at the office ( running 1.4.13) from home ( 1.6.0 beta7) over iax. 1 out of 3 times the call breaks up, but only in the home - office direction. office - home always sounds good. If it were a poor internet connection, I'd expect both sides of the conversation to be poor. Not surprisingly, each side can ping the other in the same time - 25-30ms. Both servers have the iax jitterbuffer on. I could always use a lower bit-rate codec ( now using mu-law ), but I don't see how it could be a one way bit-rate issue. Any suggestions appreciated. sean ___ Sean, Broadband connections are almost always asynchronous, which means your download speed is considerably higher than upload speed. With some or our remote workers they were getting 1.5 Mbps download but only 125 Kbps upload speed! We ended up having to upgrade their connection to a business class connection, but upload speed was still only ½ of the download speed. You can check your speed both directions with a speed test from a site such as: http://www.speakeasy.net/speedtest/ HTH I have a dsl 3m/512k line. speedtest shows 2500/400. I'm not uploading anything. I don't use my home machine as an open server. So 400k should be plenty for one voip connection and the miscellaneous chirps. sean ___ -- 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] how do I set callerid for incoming iax?
Using 1.6-rc8.
In iax.conf on the calling box, I have:
[iax-out]
.
callerid = sean 447
I even also put the same on called box.
But I can't seem to set the callerid:
exten =_NXX,1,Answer()
exten =_NXX,n,NoOp(${CALLERID(num)})
Answer(IAX2/iax-in-7, ) in new stack
NoOp(IAX2/iax-in-7, ) in new stack
So how do I set callerid?
sean
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Re: [asterisk-users] how do I set callerid for incoming iax?
Tilghman Lesher wrote:
On Tuesday 06 May 2008 12:45:42 sean darcy wrote:
Using 1.6-rc8.
In iax.conf on the calling box, I have:
[iax-out]
.
callerid = sean 447
I even also put the same on called box.
But I can't seem to set the callerid:
exten =_NXX,1,Answer()
exten =_NXX,n,NoOp(${CALLERID(num)})
Answer(IAX2/iax-in-7, ) in new stack
NoOp(IAX2/iax-in-7, ) in new stack
So how do I set callerid?
iax-out != iax-in
So??
On the calling box,
[iax-out]
type=friend
username=iax-in
secret=password
peercontext=longdistance ; which also does extensions
host=
qualify=yes
trunk=yes
callerid = sean 447
On the called - receiving - box:
[iax-in]
type=friend
username=iax-in
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid = sean 447
and then the cli shows:
-- Executing [EMAIL PROTECTED]:1] Answer(IAX2/iax-in-1, ) in
new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(IAX2/iax-in-1, )
in new stack
I must be missing something. The name of the iax.conf context matters
somehow?
Thanks for any help.
sean
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[asterisk-users] dialplan syntax error: need new eyes
I'm trying to set the outgoing caller id to the DID number, but only if
the extension is greater than 140. MAINSTUB is simply the first 7 digits
of the main number. sip.conf sets the CALLERID(num) to the extension.
exten =_1NXXNXX,n,Set(CALLERID(num)=${MAINSTUB}${CALLERID(num)})
works. But I want to set the caller id to the main number unless the
extension is 141 or higher.
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
ast_yyerror(): syntax error: syntax error, unexpected '', expecting
$end; Input:
140
I've counted my parens, checked IF syntax, and now need some new eyes to
look at this.
Thanks.
sean
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Re: [asterisk-users] dialplan syntax error: need new eyes
Barry Miller wrote:
On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
Change IF ( to IF(.
Thanks for the response.
Tried it this way:
exten =_1NXXNXX,n,Set(CALLERID(num) = ${IF($[ ${CALLERID(num)}
140] ? $
{MAINSTUB}${CALLERID(num)} : ${MAINNUMBER})})
Same result.
sean
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Re: [asterisk-users] dialplan syntax error: need new eyes
Barry Miller wrote:
On Sat, May 24, 2008 at 12:01:50AM -0400, sean darcy wrote:
Barry Miller wrote:
On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
Change IF ( to IF(.
Same result.
Sorry. This time I actually tested it. *After* de-spacing the = ,
exten = test,n,Set(CALLERID(num)=${IF( $[${CALLERID(num)} 140] ?
${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
exten = test,n,NoOp(${CALLERID(num)})
behaved properly. At least with 1.4.19.1.
I cut and pasted that, and got the same error. I'm still at 1.4.13. I'm
also testing with a blank callerid. If you could test with a blank
callerid, I'd appreciate it, but it looks like I need to upgrade.
FWIW, every time I try to use
whitespace to make a dialplan more readable, it jumps up and bites me.
Again, sorry for jumping in with an untested response.
If you hadn't responded, tested or not, I'd still be going crazy staring
at this.
Thanks for all your help.
sean
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Re: [asterisk-users] Help! - Double NAT issue
Try this. It WFM: localnet=10.0.0.0/255.255.255.0 nat = yes stunaddr = stun.ekiga.net ; or some other stun server, e.g.: foo.stun.com:3478 externrefresh = 15 ___ -- 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] beta9: how to set callerid on incoming iax?
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first stanza)
exten =_1NXXNXX,n,Dial(Zap/g0/${EXTEN})
exten =_1NXXNXX,n,Congestion()
exten =_1NXXNXX,n,Busy()
exten =_1NXXNXX,n,Hangup()
from cli:
Executing [EMAIL PROTECTED]:1] Answer(IAX2/nhi-2, ) in new stack
Executing [EMAIL PROTECTED]:2] NoOp(IAX2/nhi-2, first stanza) in
new stack
Executing [EMAIL PROTECTED]:4] Dial(IAX2/nhi-2, Zap/g0/xxx) in new stack
So extensions.conf see the _channel_ as the callerid(num). Is this a
bug, or am I messing something up?
sean
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Re: [asterisk-users] beta9: how to set callerid on incoming iax?
sean darcy wrote:
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first stanza)
exten =_1NXXNXX,n,Dial(Zap/g0/${EXTEN})
exten =_1NXXNXX,n,Congestion()
exten =_1NXXNXX,n,Busy()
exten =_1NXXNXX,n,Hangup()
from cli:
Executing [EMAIL PROTECTED]:1] Answer(IAX2/nhi-2, ) in new stack
Executing [EMAIL PROTECTED]:2] NoOp(IAX2/nhi-2, first stanza) in
new stack
Executing [EMAIL PROTECTED]:4] Dial(IAX2/nhi-2, Zap/g0/xxx) in new stack
So extensions.conf see the _channel_ as the callerid(num). Is this a
bug, or am I messing something up?
No not the channel, it's the blank before first stanza. Why's it blank?
sean
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Re: [asterisk-users] beta9: how to set callerid on incoming iax?
sean darcy wrote:
sean darcy wrote:
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first stanza)
exten =_1NXXNXX,n,Dial(Zap/g0/${EXTEN})
exten =_1NXXNXX,n,Congestion()
exten =_1NXXNXX,n,Busy()
exten =_1NXXNXX,n,Hangup()
from cli:
Executing [EMAIL PROTECTED]:1] Answer(IAX2/nhi-2, ) in new stack
Executing [EMAIL PROTECTED]:2] NoOp(IAX2/nhi-2, first stanza) in
new stack
Executing [EMAIL PROTECTED]:4] Dial(IAX2/nhi-2, Zap/g0/xxx) in new stack
So extensions.conf see the _channel_ as the callerid(num). Is this a
bug, or am I messing something up?
No not the channel, it's the blank before first stanza. Why's it blank?
So I set up ten special iax-in* contexts in extensions.conf, which set
callerid and then goto [longdistance]. Seems a weird way to do it, but
it works.
sean
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Re: [asterisk-users] beta9: how to set callerid on incoming iax?
sean darcy wrote:
sean darcy wrote:
sean darcy wrote:
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first stanza)
exten =_1NXXNXX,n,Dial(Zap/g0/${EXTEN})
exten =_1NXXNXX,n,Congestion()
exten =_1NXXNXX,n,Busy()
exten =_1NXXNXX,n,Hangup()
from cli:
Executing [EMAIL PROTECTED]:1] Answer(IAX2/nhi-2, ) in new stack
Executing [EMAIL PROTECTED]:2] NoOp(IAX2/nhi-2, first stanza) in
new stack
Executing [EMAIL PROTECTED]:4] Dial(IAX2/nhi-2, Zap/g0/xxx) in new stack
So extensions.conf see the _channel_ as the callerid(num). Is this a
bug, or am I messing something up?
No not the channel, it's the blank before first stanza. Why's it blank?
So I set up ten special iax-in* contexts in extensions.conf, which set
callerid and then goto [longdistance]. Seems a weird way to do it, but
it works.
sean
We're switching at least some to sip for better call quality. There the
callerid set in sip.conf does carry over into the dialplan.
sean
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Re: [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
Great. But I'm still a little confused. Does zaptel 1.4.12 work with asterisk-1.6.0-rc4? It looks like we first upgrade to zaptel 1.4.12, and then to dahdi. We can go back to this release of zaptel if we have problems with dahdi. Or if we go back to zaptel, do we go back to 1.6.0-beta9 also? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] conf files for dahdi
upgrading from zaptel to dahdi, with a TDM400P: Is /etc/dahdi/system.conf the same as /etc/zaptel.conf? As I read the system.conf.sample, no echo canceller need be specified if there's a hardware ec. Can I just rename zaptel.conf? What about zapata.conf? Is this just renamed /etc/asterisk/chan_dahdi.conf? Or zapata-channels.conf? Or just left alone? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI FAQ not up. Anyplace else?
http://www.asterisk.org/zaptel-to-dahdi is empty. Is there anyplace else besides the README's and Upgrade.txt's for config info on updating? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi tdm400p: no luck
As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan = 1 ; Telephone attached to port 1 dahdichan = 2 ; Telephone attached to port 2 [pstn] context=incoming-pstn-line ; Incoming calls go to [incoming-pstn-line] in extensions.conf signalling=fxs_ks ; fxs_ks Use FXS signalling for an FXO channel faxdetect=incoming busydetect=yes dahdichan = 4 ; PSTN attached to port 4 dmesg: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.0.0-rc3 ACPI: PCI Interrupt Link [APC1] enabled at IRQ 16 ACPI: PCI Interrupt :01:05.0[A] - Link [APC1] - GSI 16 (level, low) - IRQ 16 PCI: Setting latency timer of device :01:05.0 to 64 Freshmaker version: 73 Freshmaker passed register test Clocksource tsc unstable (delta = -71426924 ns) Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) INFO-xpp: revision trunk-r6056 MAX_XPDS=64 (8*8) INFO-xpp: FEATURE: without BRISTUFF support INFO-xpp: FEATURE: with PROTOCOL_DEBUG INFO-xpp: FEATURE: with sync_tick() from DAHDI INFO-xpp_usb: revision trunk-r6056 usbcore: registered new interface driver xpp_usb dahdi: Registered tone zone 0 (United States / North America) dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM400P REV I Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV I location=PCI Bus 01 Slot 06 basechan=1 totchans=4 irq=16 type=analog port=1,FXS port=2,FXS port=3,none port=4,FXO CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) But if I dial in, no dial tone, nothing on the cli. And: *CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefaultdefault In Service Tried dahdi_genconf. No help. Reverted now. Any help appreciated. seam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi tdm400p: no luck
Tzafrir Cohen wrote: On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote: As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 echocancel? I thought that if you had hardware echocancel ( TDM400P does, doesn't it? ), setting the software echocanceller was irrelevant. In any event, isn't mg2 the deefault? I'll take the system down and change this, and dahdichan to 1,2 later today, though again that wouldn't explain the lack of call pickup on the _external_ line. show daahdi channels shows _no_ channels. ( sigh) And, I'm using 1.6.0-rc4. Thanks for the quick response. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi tdm400p: no luck
Tzafrir Cohen wrote: On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote: Tzafrir Cohen wrote: On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote: As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 echocancel? I thought that if you had hardware echocancel ( TDM400P does, doesn't it? ), TDM400P doesn't. Do you mean TDM410P? setting the software echocanceller was irrelevant. In any event, isn't mg2 the deefault? No. You may have that impression from the configuration generated by dahdi_genconf that adds it as a default (that is: generates an explicit echocancel line for each channel) due to this limitation. That may change in the future if system.conf will grow up its own default echo canceller. I'll take the system down and change this, and dahdichan to 1,2 later today, though again that wouldn't explain the lack of call pickup on the _external_ line. show daahdi channels shows _no_ channels. ( sigh) And this still does not explain why you have not posted the output of: cat /proc/dahdi/* ;-) And, I'm using 1.6.0-rc4. I've got 1.6.0-rc4 up again. cat /proc/dahdi/* Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) 1 WCTDM/4/0 FXOKS 2 WCTDM/4/1 FXOKS 3 WCTDM/4/2 4 WCTDM/4/3 FXSKS and dahdi_cfg seems to have worked: dahdi_cfg -vv DAHDI Tools Version - 2.0.0-rc2 DAHDI Version: 2.0.0-rc3 Echo Canceller(s): Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 3 channels to configure. Changing signalling on channel 1 from Unused to FXO Kewlstart Changing signalling on channel 2 from Unused to FXO Kewlstart Changing signalling on channel 4 from Unused to FXS Kewlstart but still no luck. No dial tone for the internal phones, no answer on pstn. *CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) *CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefaultdefault In Service *CLI dahdi show channel 1 Unable to find given channel 1 Command 'dahdi show channel 1' failed. cat /etc/dahdi/system.conf # note change in fxo_ks and fx2_ks. 1 2 are internal, 4 is extension fxoks=1,2 fxsks=4 loadzone= us defaultzone = us BTW, this file is sometimes referred to as dahdi.conf - to keep us on our toes. and what is the comment sign ; or # ? cat /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no group=1 callgroup=1 pickupgroup=1 callprogress=yes progzone=us tonezone = 0 ; 0 is US jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; DAHDI channel. Defaults to no. An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The DAHDI channel can't accept jitter, ; thus an enabled jitterbuffer on the receive DAHDI side will always ; be used if the sendi [home-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=auto ; fxo_ks Use FXO signalling for an FXS channel - as set in sytem.conf.conf ;channel = 1 ; Telephone attached to port 1 ;channel = 2 ; Telephone attached to port 2 dahdichan = 1,2 [pstn] context=incoming-pstn-line ; Incoming calls go to [incoming-pstn-line] in extensions.conf signalling=auto ; fxs_ks Use FXS signalling for an FXO channel - use as set in system.conf faxdetect=incoming busydetect=yes ;channel = 4 dahdichan = 4 ; PSTN attached to port 4 Thanks for any help. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi tdm400p: no luck
Tzafrir Cohen wrote: On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote: Tzafrir Cohen wrote: On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote: Tzafrir Cohen wrote: On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote: As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 echocancel? I thought that if you had hardware echocancel ( TDM400P does, doesn't it? ), TDM400P doesn't. Do you mean TDM410P? setting the software echocanceller was irrelevant. In any event, isn't mg2 the deefault? No. You may have that impression from the configuration generated by dahdi_genconf that adds it as a default (that is: generates an explicit echocancel line for each channel) due to this limitation. That may change in the future if system.conf will grow up its own default echo canceller. I'll take the system down and change this, and dahdichan to 1,2 later today, though again that wouldn't explain the lack of call pickup on the _external_ line. show daahdi channels shows _no_ channels. ( sigh) And this still does not explain why you have not posted the output of: cat /proc/dahdi/* ;-) And, I'm using 1.6.0-rc4. I've got 1.6.0-rc4 up again. cat /proc/dahdi/* Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) 1 WCTDM/4/0 FXOKS 2 WCTDM/4/1 FXOKS 3 WCTDM/4/2 4 WCTDM/4/3 FXSKS and dahdi_cfg seems to have worked: dahdi_cfg -vv DAHDI Tools Version - 2.0.0-rc2 DAHDI Version: 2.0.0-rc3 Echo Canceller(s): Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 3 channels to configure. Changing signalling on channel 1 from Unused to FXO Kewlstart Changing signalling on channel 2 from Unused to FXO Kewlstart Changing signalling on channel 4 from Unused to FXS Kewlstart but still no luck. No dial tone for the internal phones, no answer on pstn. *CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) *CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefaultdefault In Service *CLI dahdi show channel 1 Unable to find given channel 1 Command 'dahdi show channel 1' failed. cat /etc/dahdi/system.conf # note change in fxo_ks and fx2_ks. 1 2 are internal, 4 is extension fxoks=1,2 fxsks=4 loadzone= us defaultzone = us BTW, this file is sometimes referred to as dahdi.conf - to keep us on our toes. and what is the comment sign ; or # ? cat /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no group=1 callgroup=1 pickupgroup=1 callprogress=yes progzone=us tonezone = 0 ; 0 is US jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; DAHDI channel. Defaults to no. An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The DAHDI channel can't accept jitter, ; thus an enabled jitterbuffer on the receive DAHDI side will always ; be used if the sendi [home-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=auto ; fxo_ks Use FXO signalling for an FXS channel - as set in sytem.conf.conf ;channel = 1 ; Telephone attached to port 1 ;channel = 2 ; Telephone attached to port 2 dahdichan = 1,2 [pstn] context=incoming-pstn-line ; Incoming calls go to [incoming-pstn-line] in extensions.conf signalling=auto ; fxs_ks Use FXS signalling for an FXO channel - use as set in system.conf faxdetect=incoming busydetect=yes ;channel = 4 dahdichan = 4 ; PSTN attached to port 4 Looks OK. What messages do you get when you run in the CLI: dahdi restart dahdi restart Destroying channels and reloading DAHDI configuration. Initial softhangup of all DAHDI channels complete. Final softhangup of all DAHDI channels complete. == Unregistered channel -2 == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found ___ -- Bandwidth and Colocation Provided by http
Re: [asterisk-users] dahdi tdm400p: no luck
sean darcy wrote: Tzafrir Cohen wrote: On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote: Tzafrir Cohen wrote: On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote: Tzafrir Cohen wrote: On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote: As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 echocancel? I thought that if you had hardware echocancel ( TDM400P does, doesn't it? ), TDM400P doesn't. Do you mean TDM410P? setting the software echocanceller was irrelevant. In any event, isn't mg2 the deefault? No. You may have that impression from the configuration generated by dahdi_genconf that adds it as a default (that is: generates an explicit echocancel line for each channel) due to this limitation. That may change in the future if system.conf will grow up its own default echo canceller. I'll take the system down and change this, and dahdichan to 1,2 later today, though again that wouldn't explain the lack of call pickup on the _external_ line. show daahdi channels shows _no_ channels. ( sigh) And this still does not explain why you have not posted the output of: cat /proc/dahdi/* ;-) And, I'm using 1.6.0-rc4. I've got 1.6.0-rc4 up again. cat /proc/dahdi/* Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) 1 WCTDM/4/0 FXOKS 2 WCTDM/4/1 FXOKS 3 WCTDM/4/2 4 WCTDM/4/3 FXSKS and dahdi_cfg seems to have worked: dahdi_cfg -vv DAHDI Tools Version - 2.0.0-rc2 DAHDI Version: 2.0.0-rc3 Echo Canceller(s): Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 3 channels to configure. Changing signalling on channel 1 from Unused to FXO Kewlstart Changing signalling on channel 2 from Unused to FXO Kewlstart Changing signalling on channel 4 from Unused to FXS Kewlstart but still no luck. No dial tone for the internal phones, no answer on pstn. *CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) *CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefaultdefault In Service *CLI dahdi show channel 1 Unable to find given channel 1 Command 'dahdi show channel 1' failed. cat /etc/dahdi/system.conf # note change in fxo_ks and fx2_ks. 1 2 are internal, 4 is extension fxoks=1,2 fxsks=4 loadzone= us defaultzone = us BTW, this file is sometimes referred to as dahdi.conf - to keep us on our toes. and what is the comment sign ; or # ? cat /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no group=1 callgroup=1 pickupgroup=1 callprogress=yes progzone=us tonezone = 0 ; 0 is US jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; DAHDI channel. Defaults to no. An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The DAHDI channel can't accept jitter, ; thus an enabled jitterbuffer on the receive DAHDI side will always ; be used if the sendi [home-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=auto ; fxo_ks Use FXO signalling for an FXS channel - as set in sytem.conf.conf ;channel = 1 ; Telephone attached to port 1 ;channel = 2 ; Telephone attached to port 2 dahdichan = 1,2 [pstn] context=incoming-pstn-line ; Incoming calls go to [incoming-pstn-line] in extensions.conf signalling=auto ; fxs_ks Use FXS signalling for an FXO channel - use as set in system.conf faxdetect=incoming busydetect=yes ;channel = 4 dahdichan = 4 ; PSTN attached to port 4 Looks OK. What messages do you get when you run in the CLI: dahdi restart dahdi restart Destroying channels and reloading DAHDI configuration. Initial softhangup of all DAHDI channels complete. Final softhangup of all DAHDI channels complete. == Unregistered channel -2 == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found on starting, CLI shows: ERROR[4384]: codec_dahdi.c:399 find_transcoders
Re: [asterisk-users] dahdi tdm400p: no luck
Matt Gibson wrote: I noticed one thing, /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 echocancel? Tzafrir mentioned it earlier, but it may have gotten lost on the thread. I was having problems with Dahdi until I added echocancel to our system.conf, could this be your problem? As soon as we added the echocanceller we were good to go. echocanceller=mg2,1-4 Thanks, Matt G I've also tried it with echocancel on: system.conf: fxoks=1,2 # internal phones fxsks=4 # pstn loadzone= us defaultzone = us echocanceller=mg2,1,2,4 which seemed to take: dahdi_cfg -vvv DAHDI Tools Version - 2.0.0-rc2 DAHDI Version: SVN-trunk-r4865 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 3 channels to configure. Changing signalling on channel 1 from Unused to FXO Kewlstart Setting echocan for channel 1 to mg2 Changing signalling on channel 2 from Unused to FXO Kewlstart Setting echocan for channel 2 to mg2 Changing signalling on channel 4 from Unused to FXS Kewlstart Setting echocan for channel 4 to mg2 [EMAIL PROTECTED] trunk]# cat /proc/dahdi/1 Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) 1 WCTDM/4/0 FXOKS (EC: MG2) 2 WCTDM/4/1 FXOKS (EC: MG2) 3 WCTDM/4/2 4 WCTDM/4/3 FXSKS (EC: MG2) But had the same result. asterisk doesn't see any channels, no dialtone, etc. It seems to me the dahdi driver works. For some reason, however, chan_dahdi doesn't see the channels the driver set up. Anybody else using TDM400P with dahdi and rc4? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi tdm400p: no luck
Matt Gibson wrote: It seems to me the dahdi driver works. For some reason, however, chan_dahdi doesn't see the channels the driver set up. Anybody else using TDM400P with dahdi and rc4? Hi Sean, Not sure if it matters, but we're using 2.0R3, noticed you're on 2.0R2 Unfortunately, I don't have an actual analogue phone here to test with, but everything LOOKS okay on our system, here's the relevant info: [EMAIL PROTECTED]:~# dahdi_cfg -vv DAHDI Tools Version - 2.0.0-rc2 DAHDI Version: 2.0.0-rc3 Echo Canceller(s): MG2 Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 4 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 [EMAIL PROTECTED]:~# cat /proc/dahdi/1 Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 FXOKS (EC: MG2) 2 WCTDM/4/1 FXOKS (EC: MG2) 3 WCTDM/4/2 FXOKS (EC: MG2) 4 WCTDM/4/3 FXSKS RED (EC: MG2) Connected to Asterisk 1.6.0-rc4 currently running on pbx (pid = 5275) pbx*CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV E/F Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) pbx*CLI dahdi show version DAHDI Version: 2.0.0-rc3 Echo Canceller: MG2 I'm also using the dahdi-linux rc3 tar ball. It's tools that are rc2: dahdi_cfg -vv DAHDI Tools Version - 2.0.0-rc2 DAHDI Version: 2.0.0-rc3 We are also on TDM400P, and we see the following with dhadi show channels *CLI dahdi show channels Chan Extension Context Language MOH InterpretBlocked State As I said we have no analogue connectivity to test with, so it could be working, or could not be for us :/ I'll see if I can dig up a phone to test with. That'd be great. It would test if it's my setup, or an issue with dahdi_chan. Thanks jay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi tdm400p: no luck - filed BUG
Tzafrir Cohen wrote: On Fri, Sep 05, 2008 at 08:39:16PM -0400, sean darcy wrote: sean darcy wrote: Tzafrir Cohen wrote: On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote: Tzafrir Cohen wrote: What messages do you get when you run in the CLI: dahdi restart dahdi restart Destroying channels and reloading DAHDI configuration. Initial softhangup of all DAHDI channels complete. Final softhangup of all DAHDI channels complete. == Unregistered channel -2 == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found So is there a problem with using dahdichan? The section does not get parsed or ignored for whatever reason? on starting, CLI shows: ERROR[4384]: codec_dahdi.c:399 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory That codec_dahdi, not chan_dahdi . You should worry about that if you have a transcoder card, which is not the case for you. Tried dahdi-linux-rc4 and asterisk-1.6.0-rc5. Same problem. Filed bug: http://bugs.digium.com/view.php?id=13443 sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi tdm400p: no luck - filed BUG
sean darcy wrote: Filed bug: http://bugs.digium.com/view.php?id=13443 sean For any one else who has this problem: don't use user-defined sections, i.e. [pstn] If you use dahdichan = x , chan_dahdi will only use the _last_ dahdichan statement for channel(s). channels = works as expected. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] what codec is sip using?
If you use iax, the console will tell you what codec is being used. But for sip, nothing is shown. With sip debug on, I get: Capabilities: us - 0x130e (gsm|ulaw|alaw|g729|speex|g722), peer - audio=0x100e (gsm|ulaw|alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100e (gsm|ulaw|alaw|g722) but I don't see anything that shows which codec was used. How do I find out? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what codec is sip using?
David Gibbons wrote: Sean, Try 'sip show channels' or 'sip show channel channelid' for the drill down. I believe the codec in use will be displayed with either command. Dave Thanks that worked. Now how do I get it show the codec when I'm not at the CLI? sean From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of sean darcy [EMAIL PROTECTED] Sent: Thursday, September 18, 2008 10:49 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] what codec is sip using? If you use iax, the console will tell you what codec is being used. But for sip, nothing is shown. With sip debug on, I get: Capabilities: us - 0x130e (gsm|ulaw|alaw|g729|speex|g722), peer - audio=0x100e (gsm|ulaw|alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100e (gsm|ulaw|alaw|g722) but I don't see anything that shows which codec was used. How do I find out? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what codec is sip using?
Alex Balashov wrote: sean darcy wrote: David Gibbons wrote: Sean, Try 'sip show channels' or 'sip show channel channelid' for the drill down. I believe the codec in use will be displayed with either command. Dave Thanks that worked. Now how do I get it show the codec when I'm not at the CLI? Show it where, if not the CLI? sorry, I wasn't clear. I'd like it to show up on the CLI without having to enter the sip show channel command at the time of the call. Just as iax does. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] requested special control 20 ??
I'm using Teliax, and every incoming call has: Executing [EMAIL PROTECTED]:2] Answer(IAX2/usrname-14376, ) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(IAX2/usrname-14376, DAHDI/1,60) in new stack -- Called 1 -- DAHDI/1-1 is ringing -- IAX2/usrname-14376 requested special control 20, passing it to DAHDI/1-1 -- IAX2/usrname-14376 requested special control 20, passing it to DAHDI/1-1 -- DAHDI/1-1 is ringing It all seems to work OK, but what's requesting special control 20 all about? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((
Remco Barendse wrote: The information (or lack of it) on upgrading from zaptel to that @*^QW%^%!!! dahdi is very frustrating. I cannot find anything on how to uninstall zaptel, i found an earlier post to this list which suggested make uninstall and make remove in the zaptel directory which just generates errors and does nothing (on zaptel 12.1). Then i install dahdi-linux and dahdi-tools and i want to start configuring it, so i am trying dahdi_genconf like the docs suggested which generates this really helpful error message : /usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No such file or directory Also the config files and everything are much more complicated for dahdi than they were for zaptel There was some nice documentation and examples on how to get started with configuring certain devices with zaptel on the digium page, for my TDM11B they only mention zaptel. Did anyone even try this? It'll work. But it's not easy. I didn't find dahdi_genconf helpful. Post your /etc/dahdi/system.conf ( the analogue of zaptel.conf ) and /etc/asterisk/chan_dahdi.conf ( analogue of zapata.conf ). With some help, you'll fix this. sean ___ -- 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] 1.6.0.1 ??
In download dated 10/9. Bug fix? Mistake? sean ___ -- 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] setup for fax machine
Becasue of all the issues with fax over voip, we want to use pstn for our fax machine, but not dedicate a line just to fax. I'm thinking of having asterisk answer the pstn line, check for fax tones, and route appropriately. In zapata ( chan_dahdi ) set faxdetect=incoming then the dial plan would have [incoming-pstn] exten = fax,1,Dial(DAHDI/1) ; the fax machine exten = fax,2,Hangup() exten = s,1,Answer() exten = s,2,Dial(DAHDI/2) ; internal extension . Would this work? I'll need another TDM410 card to do this, so I'd like some reassurance before I go purchase it. sean ___ -- 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] a little regex help needed
I'm trying to set the callerid(name) to Office for all calls from the
main office.
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
The main office callerid's are all 212 457 11xx. But this statement
seems to match everything, including callerid(num)=
What I'd expect is a callerid(num) of 2124571123 to generate an if test
of [02124571123 == 021245711*] or TRUE.
But I've messed up the regex statement somehow.
Thanks for any help.
sean
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Re: [asterisk-users] QoS VoIP
Anael DIAZ wrote: Hi! I have some problem in my asterisk 1.4.2, I've installed it on centOS 5.2 and this didn't accept voip QoS and can't route the packets having voip QoS. So I should change voip packets to be routing with centOS. I want to use iproute2 but i don't what to do after installing iproute2. Anyone could help me please? ___ -- 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 This stuff is pure voodoo. I've found very little good specific instruction. I put this into rc.local to set up QoS. I'm not sure I understood it then, and I'm sure I don't understand it now, but it may be useful to you. I also put the various tos stuff in sip.conf, etc. cat tos.local ## eth1 is the external interface ## remove the queues EXTIF=eth1 tc qdisc del dev $EXTIF root ## This is to set up QoS for voip - specifically iax. ## from http://www.howtoforge.com/voip_qos_traffic_shaping_iproute2_asterisk ## ethx is the *external* port tc qdisc add dev $EXTIF root handle 1: prio priomap 2 2 2 2 2 2 2 2 1 1 1 1 1 1 1 0 tc filter add dev $EXTIF protocol ip parent 1: prio 1 u32 match ip dport 4569 0x flowid 1:1 tc filter add dev $EXTIF protocol ip parent 1: prio 1 u32 match ip sport 4569 0x flowid 1:1 tc filter add dev $EXTIF protocol ip parent 1: prio 1 u32 match ip tos 0x10 0xff flowid 1:2 Please post anything you do find. Good luck. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a little regex help needed
Jared Smith wrote:
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
[snip]
What I'd expect is a callerid(num) of 2124571123 to generate an if test
of [02124571123 == 021245711*] or TRUE.
But I've messed up the regex statement somehow.
In regular expressions, the * means zero or more of the preceding
character, so the way you have that written means 021245711 and zero or
more 1s. What you want instead is 021245711.*, which means
021245711 followed by at least on other character.
Hopefully that sets you on the right path. Don't forget that Asterisk
has two regex operators that can be used in expressions as well...
they're the ':' and '~' operators.
OK. So I changed the * to .. , like so:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711..)} ] ? Office:${CALLERID(name)} )})
which I would expect to mean 021245711 followed by two other characters.
It still matches a blank callerid(num).
sean
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Re: [asterisk-users] a little regex help needed
Atis Lezdins wrote:
On Mon, Oct 20, 2008 at 9:20 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
[snip]
What I'd expect is a callerid(num) of 2124571123 to generate an if test
of [02124571123 == 021245711*] or TRUE.
This is not the correct way to use regular expressions. Regular
expression is matched to data withing REGEX function, and it just
returns match/don't match.
Here's description
REGEX(regular expression data)
[Synopsis]
Regular Expression
[Description]
Returns 1 if data matches regular expression, or 0 otherwise.
Please note that the space following the double quotes separating the
regex from the data
is optional and if present, is skipped. If a space is desired at the
beginning of the data,
then put two spaces there; the second will not be skipped.
So, it would be something like:
${REGEX(21245711.* ${CALLERID(num)})}
But I've messed up the regex statement somehow.
In regular expressions, the * means zero or more of the preceding
character, so the way you have that written means 021245711 and zero or
more 1s. What you want instead is 021245711.*, which means
021245711 followed by at least on other character.
correction - 021245711.* would match also 021245711 as * allows zero
or more and dot means any character.
Hopefully that sets you on the right path. Don't forget that Asterisk
has two regex operators that can be used in expressions as well...
they're the ':' and '~' operators.
I wonder what are those used for? Never heard of that.
Are you really sure you need regular expressions there? Asterisk has
it's own number pattern matching, as it's much easier to read, and
would allow easy adding/removing some specific masks. Here's one
sample:
[main]
..
exten = s,n,GoSub(callerid-update,${CALLERID(num)},1)
..
[callerid-update]
exten = 021245711XX,1,Set(CALLERID(name)=Office: ${CALLERID(num)});
exten = 021245711XX,2,Return();
exten = _X.,1,Return();
exten = i,1,Return(); // just for safety :)
Exactly where I was trying to go. I was thinking a little differently
though:
[any-incoming-context]
exten = s,1,Answer()
exten = s,2,Gosub(set-callerid-name,1,1)
exten = s,3,Dial(..
[set-callerid-name]
exten=1,1,Set(CALLERID(name)=${IF($[0${CALLERID(num)} = ...
exten=1,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} = ...
.
.
exten = 1,n,Return()
which seemed easier ( and easier to read) since I didn't have to insert
Return()'s every other line.
sean
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Re: [asterisk-users] a little regex help needed
On Mon, Oct 20, 2008 at 7:38 PM, sean darcy [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Mon, Oct 20, 2008 at 9:20 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
[snip]
What I'd expect is a callerid(num) of 2124571123 to generate an if test
of [02124571123 == 021245711*] or TRUE.
This is not the correct way to use regular expressions. Regular
expression is matched to data withing REGEX function, and it just
returns match/don't match.
Here's description
REGEX(regular expression data)
[Synopsis]
Regular Expression
[Description]
Returns 1 if data matches regular expression, or 0 otherwise.
Please note that the space following the double quotes separating the
regex from the data
is optional and if present, is skipped. If a space is desired at the
beginning of the data,
then put two spaces there; the second will not be skipped.
So, it would be something like:
${REGEX(21245711.* ${CALLERID(num)})}
But I've messed up the regex statement somehow.
In regular expressions, the * means zero or more of the preceding
character, so the way you have that written means 021245711 and zero or
more 1s. What you want instead is 021245711.*, which means
021245711 followed by at least on other character.
correction - 021245711.* would match also 021245711 as * allows zero
or more and dot means any character.
Hopefully that sets you on the right path. Don't forget that Asterisk
has two regex operators that can be used in expressions as well...
they're the ':' and '~' operators.
I wonder what are those used for? Never heard of that.
Are you really sure you need regular expressions there? Asterisk has
it's own number pattern matching, as it's much easier to read, and
would allow easy adding/removing some specific masks. Here's one
sample:
[main]
..
exten = s,n,GoSub(callerid-update,${CALLERID(num)},1)
..
[callerid-update]
exten = 021245711XX,1,Set(CALLERID(name)=Office: ${CALLERID(num)});
exten = 021245711XX,2,Return();
exten = _X.,1,Return();
exten = i,1,Return(); // just for safety :)
Exactly where I was trying to go. I was thinking a little differently
though:
[any-incoming-context]
exten = s,1,Answer()
exten = s,2,Gosub(set-callerid-name,1,1)
exten = s,3,Dial(..
[set-callerid-name]
exten=1,1,Set(CALLERID(name)=${IF($[0${CALLERID(num)} = ...
exten=1,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} = ...
.
.
exten = 1,n,Return()
which seemed easier ( and easier to read) since I didn't have to insert
Return()'s every other line.
But, as I think about it ( instead of just hitting Send),can yours
does work without the returns?
exten = s,2,Gosub(set-callerid-name,${CALLERID(num),1)
[set-callerid-name]
exten=21245711XX ,1,Set(CALLERID(name)=Office
exten=someother-id-num,1,Set(CALLERID(name)=...
.
.
exten = What's the extension here?,2,Return()
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[asterisk-users] set(CALLERID(name) not working
I've tried to create a subroutine that sets callerid name based on number.
extensions.conf:
...
exten = s,1,Answer()
exten = s,n,GoSub(set-callerid-name,0${CALLERID(num)},1)
exten = s,n,Dial(${mainline},60)
...
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02025462677,1,Set(CALLERID(name) = Fred )
exten = _X.,2,NoOp(CALLERID: ${CALLERID(name)})
exten = _X.,3,Return()
But it doesn't work. CALLERID(name) isn't changed:
...
-- Executing [EMAIL PROTECTED]:2] Answer(IAX2/john-7775, )
in new stack
-- Executing [EMAIL PROTECTED]:4] Gosub(IAX2/john-7775,
set-callerid-name|02025462677|1) in new stack
-- Executing [EMAIL PROTECTED]:1]
Set(IAX2/john-7775, CALLERID(name) = Fred ) in new stack
-- Executing [EMAIL PROTECTED]:2]
NoOp(IAX2/john-7775, CALLERID: Cell Phone CT) in new stack
-- Executing [EMAIL PROTECTED]:3]
Return(IAX2/john-7775, ) in new stack
-- Executing [EMAIL PROTECTED]:5] Dial(IAX2/john-7775,
DAHDI/1|60) in new stack
-- Called 1
Why not? How come CALLERID(name) isn't Fred??
sean
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Re: [asterisk-users] set(CALLERID(name) not working
OCG Technical Support wrote: Take a look at smartCID at www.generationd.com This tool will set callerid based on number in a database. If not listed there, it will search 411 for reverse lookup etc. It will also let you flag calls for blocking, etc.. Interesting. It looks like more than I need, but if I can't get this working... sean ___ -- 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] set(CALLERID(name) not working
Trevor Peirce wrote:
sean darcy wrote:
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02025462677,1,Set(CALLERID(name) = Fred )
exten = _X.,2,NoOp(CALLERID: ${CALLERID(name)})
exten = _X.,3,Return()
But it doesn't work. CALLERID(name) isn't changed:
Perhaps try this:
[set-callerid-name]
exten = 02025462677,1,Set(CALLERID(name)=Fred)
Thanks for the reply.
With or without quotes, same result.
BTW, I'm using 1.4.22.
sean
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Re: [asterisk-users] set(CALLERID(name) not working
C F wrote: Who you calling? Is it a remote non PSTN phone number? Or a PSTN number? It's incoming. Both pstn and voip. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set(CALLERID(name) not working
sean darcy wrote:
I've tried to create a subroutine that sets callerid name based on number.
extensions.conf:
...
exten = s,1,Answer()
exten = s,n,GoSub(set-callerid-name,0${CALLERID(num)},1)
exten = s,n,Dial(${mainline},60)
...
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02025462677,1,Set(CALLERID(name) = Fred )
exten = _X.,2,NoOp(CALLERID: ${CALLERID(name)})
exten = _X.,3,Return()
But it doesn't work. CALLERID(name) isn't changed:
...
-- Executing [EMAIL PROTECTED]:2] Answer(IAX2/john-7775, )
in new stack
-- Executing [EMAIL PROTECTED]:4] Gosub(IAX2/john-7775,
set-callerid-name|02025462677|1) in new stack
-- Executing [EMAIL PROTECTED]:1]
Set(IAX2/john-7775, CALLERID(name) = Fred ) in new stack
-- Executing [EMAIL PROTECTED]:2]
NoOp(IAX2/john-7775, CALLERID: Cell Phone CT) in new stack
-- Executing [EMAIL PROTECTED]:3]
Return(IAX2/john-7775, ) in new stack
-- Executing [EMAIL PROTECTED]:5] Dial(IAX2/john-7775,
DAHDI/1|60) in new stack
-- Called 1
Why not? How come CALLERID(name) isn't Fred??
sean
Well I never did find out. What I did find out is that I could only set
CALLERID(name) in the context the received the call, not in the
subroutine. But if I set a dummy variable ( Set(cidname=Fred) ) in the
subroutine, I could set the callerid back in the originating context (
Set(CALLERID(name) = ${cidname} ).
Makes no sense, but there you are.
sean
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Re: [asterisk-users] set(CALLERID(name) not working
Daniel Lynes wrote: You'll need to lose the double quotation marks in the assignment: Set(CALLERID(name)=Fred) becomes: Set(CALLERID(name)=Fred) If it still doesn't work, then it means that your particular provider does not support the ability to be able to set the caller ID name, or it's receiving a corrupted copy of it. One such provider that I'm aware of for that issue is Group Telecom. They receive corrupted caller ID name information from asterisk. So, caller ID num information will work, but not caller ID name information. Tried it with and with quotes. Same result - exactly. Works with dummy variable, doesn't if set in subroutine. This is incoming. I'm setting the CID(name) based on the incoming CID(num). Nothing to do with the provider. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set(CALLERID(name) not working
Doug Lytle wrote:
sean darcy wrote:
Tried it with and with quotes. Same result - exactly. Works with dummy
variable, doesn't if set in subroutine.
It works fine for me, I use the below:
exten = 3175797960,1,Gosub(get_name,s,1)
[get_name]
;
;* Connect to local MySQL database to match-
;* against speed dial database for Caller*ID
;* name. If no match, check against Corporate
;* Database. If a match, jump to incoming
;* Context.
;
exten = s,1,MYSQL(Connect connid localhost anonymous '' speeddials)
exten = s,n,GosubIf($[${MYSQL_STATUS} = -1]?mysql_failed,s,6)
exten = s,n,MYSQL(Query resultid ${connid} SELECT name FROM
Indianapolis WHERE phone = 1${CALLERID(num)})
exten = s,n,MYSQL(Fetch fetchid ${resultid} caller.name)
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,Set(CALLERID(name)=${caller.name})
exten = s,n,Set(CALLERID(num)=${CALLERID(num)})
exten = s,n,GotoIf($[${caller.name} != ]?10:100)
exten = s,n,NoOP(${DIALSTATUS})
exten = s,n,Return
Well, it wasn't the quotes, it wasn't you can't set CALLERID(name) in a
subroutine, it's that the parser doesn't allow/get confused by/ space
around the assignment.
Set(CALLERID(name)=Fred) works
but neither
Set(CALLERID(name) = Fred)
nor
Set(CALLERID(name) =Fred)
works.
BTW, quotes don't make a difference.
And the parser doesn't complain, or make an assignment to null. There's
just no assignment despite what the CLI shows.
In any event, thanks for the lead, and when I get the time, I'm
switching to your script.
sean
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Re: [asterisk-users] set(CALLERID(name) not working
Matt Riddell wrote: On 13/11/2008 10:23 a.m., sean darcy wrote: Tried it with and with quotes. Same result - exactly. Works with dummy variable, doesn't if set in subroutine. This is incoming. I'm setting the CID(name) based on the incoming CID(num). Nothing to do with the provider. Are you maybe seeing this: http://bugs.digium.com/view.php?id=13597 Interesting. I am on 1.4.22. But this appears to be the finicky parser. sean ___ -- 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] app directory error: libc-client undefined symbol
Installing 1.4.23-rc2, I actually looked at the startup and saw this warning: WARNING[10730]: loader.c:359 load_dynamic_module: Error loading module 'app_directory.so': /usr/lib/libc-client.so.2007: undefined symbol: mm_dlog I'm running Fedora Core 9, with libc-client 2007d. googling didn't help, so what's the problem? Do I need a more recent (different) libc-client? sean ___ -- 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] docs for rxfax in 1.4 or app_fax in 1.6?
I just want to pdf and email faxes coming in over pstn on a TDM400P.
Outgoing faxes would just go out over pstn, not through asterisk.
All the voipinfo , etc, howto's are quite complicated. And most use
third party apps like Hylafax.
I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm
now using 1.4.22, but I'd go to 1.6 if it made this easier.
But I've found no docs or sample configs for either 1.4 or 1.6. In fact,
1.4.22 ( nor addons nor 1.4.23 rc2 ) have no rx{tx}fax.c files that I'd
expect.
I do have spandsp installed, FWIW.
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Re: [asterisk-users] docs for rxfax in 1.4 or app_fax in 1.6?
Philipp Kempgen wrote:
sean darcy schrieb:
I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm
now using 1.4.22, but I'd go to 1.6 if it made this easier.
But I've found no docs or sample configs for either 1.4 or 1.6. In fact,
1.4.22 ( nor addons nor 1.4.23 rc2 ) have no rx{tx}fax.c files that I'd
expect.
I do have spandsp installed, FWIW.
TxFax() / RxFax() are available as a patch for 1.4. Somewhere.
Sorry, no pointers. But certainly easy to find.
1.6 comes with SendFax() / ReceiveFax() which use SpanDSP as
well and replace txfax/rxfax.
Philipp Kempgen
Thanks for the reply.
I'll probably try ReceiveFax() if it's in 1.6. Patching outside apps
sometimes leads to version trouble.
Any docs for ReceiveFax()? Or, if I'm really lucky, an example?
sean
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[asterisk-users] 1.6.1: iax trunk needs dahdi timing ??
starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these timing modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what iax is looking for? sean ___ -- 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] 1.6.1: iax trunk needs dahdi timing ??
Russell Bryant wrote: Michiel van Baak wrote: On 20:24, Sun 14 Dec 08, sean darcy wrote: starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these timing modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what iax is looking for? If you dont have any dahdi hardware installed and configured, make sure to load dahdi_dummy. That will provide you the timers. In 1.6.1, this should not be required. It's probalby a check in the code that shouldn't be there anymore. If you post this on bugs.digium.com, I'll remove it. OK, it's http://bugs.digium.com/view.php?id=14082 BTW I do have a TDM400P with dahdi-2.1.0 installed and configured. So dahdi_dummy wouldn't help. sean ___ -- 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] 1.6.1: iax trunk needs dahdi timing ??
Russell Bryant wrote: Michiel van Baak wrote: On 20:24, Sun 14 Dec 08, sean darcy wrote: starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these timing modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what iax is looking for? If you dont have any dahdi hardware installed and configured, make sure to load dahdi_dummy. That will provide you the timers. In 1.6.1, this should not be required. It's probalby a check in the code that shouldn't be there anymore. If you post this on bugs.digium.com, I'll remove it. Thanks for fixing this so promptly. sean ___ -- 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] Setup ReceiveFax(), fax2mail, mime-construct - but now Sendmail :(
Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got this far:
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM})
exten = s,2,ReceiveFAX(${FAXFILE}.tif)
exten = s,3,Hangup()
exten=h,1,System(/usr/local/bin/fax2mail --cid-number 0${CALLERIDNUM}
--cid-name home fax --dest-name admin --dest-email ${admin_email}
-f ${FAXFILE})
which all seems work well on the CLI. No errors.
fax2mail uses mime-contruct to send the fax by sendmail. That didn't work.
No email. /var/log/maillog:
Dec 19 21:04:53 asterisk sendmail[2628]: mBH2mWvQ006043:
to=ad...@myco.com, ctladdr=r...@localhost.localdomain (0/0),
delay=2+23:16:09, xdelay=00:00:00, mailer=esmtp, pri=6640305,
relay=mx01.1and1.com., dsn=4.0.0, stat=Deferred: Connection timed out
with mx01.1and1.com.
Dec 19 21:04:53 asterisk sendmail[2628]: mBH2mWvS006043:
to=ad...@myco.com, ctladdr=r...@localhost.localdomain (0/0),
delay=2+23:16:09, xdelay=00:00:00, mailer=esmtp, pri=6640312,
relay=mx00.1and1.com., dsn=4.0.0, stat=Deferred: Connection timed out
with mx00.1and1.com.
I've avoided MTA's like sendmail for a _long_ time. So I need help.
1. Is this the right list to try to resolve this? If not, which list?
2. postfix seems to considered much easier to configure than sendmail.
Do I install postfix? If so, will this work out of the box?
3. If sendmail, what's the magic configuration?
sean
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[asterisk-users] 1.6.1-rc4: extension i not working??
I've have a simple caller id lookup on incoming:
[teliax-in]
..
exten =s,n,GoSub(set-callerid-name,0${CALLERID(num)},1)
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02135590993,1,Set(CALLERID(name)=Matthew )
...
exten = _0!,n,NoOp(CALLERID: ${CALLERID(name)})
exten = _0!,n,Return()
exten = i,1,Return() ; somebody else
Now if there's a callerid that's listed, it all works OK. If there's no
callerid, that works. But if there's an unknown callerid, I'd expect
that to go to the invalid extension - i - and Return(). But look
what happens:
-- Executing [2136398...@teliax-in:4] Gosub(IAX2/poseidon-15117,
set-callerid-name,02136990505,1) in new stack
[Dec 25 13:06:32] ERROR[26483]: app_stack.c:286 gosub_exec: Attempt to
reach a non-existent destination for gosub: (Context:set-callerid-name,
Extension:02136990505, Priority:0)
== Spawn extension (teliax-in, 2136398447, 5) exited non-zero on
'IAX2/johnfbeatty-15117'
-- Hungup 'IAX2/poseidon-15117'
-- Hungup 'DAHDI/4-1'
Is this a bug in 1.6.1, or an improper use of the i extension?
sean
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Re: [asterisk-users] IMAP Voicemail and Directory not working?
On Tue, Dec 23, 2008 at 10:13 AM, Noah Miller noahisaacmil...@gmail.com wrote: Hi Tzafrir - I'm wondering if anybody has IMAP Voicemail AND the directory working together. I haven't had any success. IMAP voicemail works fine, but when it's active, the Directory does not work. The problem seems to be with libc-client. Specifically, asterisk is not able to access the mm_dlog function. I've tried with Asterisk 1.4.22+ and 1.6.0+ using CentOS 5.2, Ubuntu 8.10 and Fedora 9. In each case, I used the native package manager to install libc-client, and in each case, after asterisk is compiled and voicemail users are configured, I get an error in the log that says this: On Ubuntu and Debian (Lenny/Sid) - apt-get source asterisk # as root / using sudo: apt-get build-dep asterisk cd asterisk-1tabtab ASTERISK_NO_DOCS=yes fakeroot debian/rules build Does it build? If so, you have a similar version of Asterisk that builds with IMAP support. I finally got this to work. For some reason, none of the packaged versions of libc-client from any of the distributions I tried support mm_dlog, which is required by the Directory app. I ended up compiling from uw-imap's source on Ubuntu, and that worked right away. On the Red Hat varieties, compiling from source worked, but I had to specify -fPIC and a few other compiler flags when building UW's c-client. For the record, if anybody needs to do this on a redhat platform: 1. Download imap-2007e (or latest version) from ftp://ftp.cac.washington.edu/imap/ 2. Unpack and compile with a make command like: make platform SSLTYPE=none EXTRACFLAGS=-DIGNORE_LOCK_EACCES_ERRORS=1 \ -I/usr/include/openssl -fPIC -fno-strict-aliasing -Wall -Wno-pointer-sign -Wno-parentheses (See the Makefile for a list of platforms - I used 'lr5' for CentOS 5.2) 3. In the asterisk source, run the configure script with the imap flag: ./configure --with-imap=/path/to/imap-source (use the base directory of the imap source - e.g. /usr/src/imap-2007e ) 4. Run make menuselect for asterisk and select IMAP_STORAGE from the Voicemail Build Options. Of course, you'll also need an appropriately configured IMAP server (for CentOS, I recommend their default choice of Dovecot). - Noah Very interesting. I tried this with 1.6.1-beta4 on Fedora 9. On startup, asterisk looks in the --with-imap folder which has just a static lib, but not a shared lib. The static lib can be installed with the uw-imap-static rpm. But even if the static lib is installed, asterisk chooses the shared lib over the static lib. So...something about how Fedora builds the shared lib screws it up. I looked at the spec file, but couldn't see anything. Does any distro have a shared lib that works? And asterisk should only use the static lib even if the shared lib is available - or at least have a configure switch that requires the static lib. sean sean sean ___ -- 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] IMAP Voicemail and Directory not working?
On Tue, Dec 30, 2008 at 1:30 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Dec 30, 2008 at 11:15:54AM -0500, sean darcy wrote: Very interesting. I tried this with 1.6.1-beta4 on Fedora 9. On startup, asterisk looks in the --with-imap folder which has just a static lib, but not a shared lib. The static lib can be installed with the uw-imap-static rpm. But even if the static lib is installed, asterisk chooses the shared lib over the static lib. What version of imap is it? -- 2007e. Same as Noah. sean ___ -- 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] 1.6.1-b4: Can't get fax2mail work from System()
On 1.6.1-beta4:
Trying to receive faxes over a pstn line. extensions.conf:
[incoming-pstn-line]
exten = fax,1,NoOp(Fax Detected)
exten = fax,2,GoTo(incoming-fax,s,1)
exten = fax,n,Hangup()
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM})
exten = s,2,ReceiveFAX(${FAXFILE}.tif)
exten = s,3,Hangup()
exten=h,1,System(/usr/local/bin/fax2mail.1.sh --dest-name Sean
--dest-email ${Sean_email} -f ${FAXFILE})
which looks like it works just fine from the cli:
-- DAHDI/2-1 is ringing
-- Redirecting DAHDI/4-1 to fax extension
-- Hungup 'DAHDI/2-1'
== Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on
'DAHDI/4-1'
-- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax
Detected) in new stack
-- Executing [...@incoming-pstn-line:2] Goto(DAHDI/4-1,
incoming-fax,s,1) in new stack
-- Goto (incoming-fax,s,1)
-- Executing [...@incoming-fax:1] Set(DAHDI/4-1,
FAXFILE=/var/spool/asterisk/fax/200901141711-0) in new stack
-- Executing [...@incoming-fax:2] ReceiveFAX(DAHDI/4-1,
/var/spool/asterisk/fax/200901141711-0.tif) in new stack
-- Executing [...@incoming-fax:3] Hangup(DAHDI/4-1, ) in new stack
== Spawn extension (incoming-fax, s, 3) exited non-zero on 'DAHDI/4-1'
-- Executing [...@incoming-fax:1] System(DAHDI/4-1,
/usr/local/bin/fax2mail.1.sh --dest-name Sean --dest-email
seandar...@gmail.com -f /var/spool/asterisk/fax/200901141711-0) in
new stack
-- Hungup 'DAHDI/4-1'
But it doesn't - no email is ever sent. BUT, if I execute the fax2mail
cmd from the terminal (pasting from the cli output) it sends the email:
/usr/local/bin/fax2mail.1.sh --dest-name Sean --dest-email
seandar...@gmail.com -f /var/spool/asterisk/fax/200901141711-0
Am I screwing up the System() command somehow? Is System() screwed up in
1.6.1?
Any clues how to debug this? I did find one relevant thread
http://asteriskforum.ru/viewtopic.php?p=15629 , which is unfortunatley
in Russian. In that thread someone figured out how to turn on DEBUG for
app_fax. How did you do that?
sean
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Re: [asterisk-users] 1.6.1-b4: Can't get fax2mail work from System()
OCG Technical Support wrote: Start with your mail log. Any errors visible? How about system log - PAMpermission errors? Thanks for the quick response. maillog shows nothing if it's executed from the System() call. Obviously maillog shows the outgoing if executed from the terminal, Nothing in syslog. asterisk is running as root. sean ___ -- 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] how to debug mime-construct with fax2mail?
I'm trying to capture faxes on 1.6.1-beta4. AFAICT, app_fax is working
OK. I'm then using fax2mail to send the fax. That wasn't working, so i
posted for help using the System() cmd, since fax2mail did work from the
command line. But now I realize it's fax2mail and mime-construct itself.
I set up a fax-test context:
[fax-test]
exten=666,1,NoOp( fax-test )
exten=666,2,System(/bin/echo this is a system
test${STRFTIME(${EPOCH},,%H%M)} /opt/system-test)
exten=666,3,System(/usr/local/bin/fax2mail.1.sh --dest-name Sean
--dest-email seandar...@gmail -f /var/spool/asterisk/fax/FAXFILE)
exten=666,n,Hangup
This works fine on the cli. And /opt/system-test captures the /bin/echo
string.
AND, the fax2mail log - /var/log/asterisk/faxlog - shows that fax2mail
was`called, and there are no errors. So it's not the System() cmd. But
the email is NOT sent.
faxlog:
fax2mail v2.3
Triggered on Thursday, January 15 2009, at 02:45 PM
Called with --dest-name Sean --dest-email seandar...@gmail -f
/var/spool/asterisk/fax/FAXFILE
CallerID number of fax sender =
CallerID name of fax sender = Someone Unknown
Fax number called = 213 666 9505
Destination name = Sean
Destination email address = seandar...@gmail
Fax file name (without .tif extension) = /var/spool/asterisk/fax/FAXFILE
Attachment format conversion = pdf
Set CallerID number of fax sender to unknown number
Fax file /var/spool/asterisk/fax/FAXFILE.tif found.
Converted /var/spool/asterisk/fax/FAXFILE.tif to
/var/spool/asterisk/fax/FAXFILE.pdf.
E-mailed file to seandar...@gmail
Removing destination file /var/spool/asterisk/fax/FAXFILE.pdf
I can run the exact same cmd from the terminal, and it works. The email
is sent. And the fax2mail log looks the same.
asterisk is running as root, I run the command at the terminal as root.
So I getting to think it's somehow mime-construct, which doesn't seem to
have some nice log around, even if run with --debug.
Any help really appreciated. I'm puzzled as hell.
sean
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Re: [asterisk-users] how to debug mime-construct with fax2mail?
Joseph L. Casale wrote: Have you tried your system stuff under su - asterisk? Once it works that way, the system() command will work. asterisk is running as root, I run the command at the terminal as root. I am guessing he doesn't even have an asterisk user. Well I do have an asterisk user, and once spent a weekend trying to run asterisk as asterisk user. But I don't see what this has to do with my problem. The System() cmd works: I can see the log from fax2mail showing it was called, and called with the arguments I expected. So System() did it's thing. What I can't figure what is why fax2mail really works from the command line, but fails to effectively call mime-construct when called from System(). I was hoping someone who has used mime-construct could show me how to debug it. It may be a permissions problem, but since both run as root it seems unlikely. In any event, being able to debug mime-construct would allow me to figure it out. sean ___ -- 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 to debug mime-construct with fax2mail?
Lyle Giese wrote: If you are running the script within Asterisk as root, then it's a path environment issue. My guess(and I run into this with cron jobs all the time) is that the path is different from the command line than the environment that the script runs under. There are times where the fix is to use the fully qualified path when calling stuff and not assume it's in the path. Lyle You are the man. If we ever meet I owe you a beer, at least one. In the fax2mail script, it just calls mime-construct without a full path. mime-construct on my box is in /usr/local/bin which must not be in the path of the environment System calls are run in. Putting in the fully qualified path made it work. Thanks again. sean ___ -- 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 to debug mime-construct with fax2mail?
OCG Technical Support wrote: If you want to email me your fixed script I'll put it up on the web site... Well I'd be pleased to have any script of mine put up on any web site, but the only thing I did was to hard wire my location of mime-construct: MimeC=/usr/local/bin/mime-construct and the changed all the calls to mime-construct to MimeC. Not very portable :( I suppose what should happen is a test if mime-construct is in the path, and then a search. But this is waay beyond my scripting prowess. sean ___ -- 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] pstn hangs up: MWI no message waiting ??
pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! -- Hungup 'DAHDI/4-1' I don't have any Message Waiting set ( or at least I don't think so.) Restarting * solves it for a while. Any suggestions? sean ___ -- 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] pstn hangs up: MWI no message waiting ??
Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! This message occurs when the pstn sends a FSK spill indicating the message waiting status of the FXO port in question. This may encoded in the caller ID indicator or may be contained in its own message spill. This is output as a NOTICE logging message. Regards, Doug Bailey I'm not sure I understand all that, but why does asterisk hang up? It means I can't receive any calls on that pstn line. AFAICS, only restarting asterisk allows calls to be received. A cron job to restart every 5 minutes? sean ___ -- 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] pstn hangs up: MWI no message waiting ??
Danny Nicholas wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. Yes, a cron job to restart zaptel would cut off any call then existing. But how would I test for it? I can imagine: exten=s,n,ExecIf(some damn thing, System(service dahdi restart)) It's the some damn thing I can't imagine. How do you test if dahdi is acting up? sean ___ -- 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] pstn hangs up: MWI no message waiting ??
Tilghman Lesher wrote: On Friday 16 January 2009 17:43:21 sean darcy wrote: Danny Nicholas wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. Yes, a cron job to restart zaptel would cut off any call then existing. But how would I test for it? I can imagine: exten=s,n,ExecIf(some damn thing, System(service dahdi restart)) It's the some damn thing I can't imagine. How do you test if dahdi is acting up? Not a service restart, but a dahdi restart. You can't restart the dahdi service without first stopping Asterisk, anyway. if [ `../asterisk-trunk/contrib/scripts/astcli core show channels | wc -l` = 3 ]; then asterisk -rx 'dahdi restart'; fi Wow. I'll try that tomorrow. Put it as the cmd right after answer(), right? Or maybe, h,1 ? Well anyway, at least I'll be able to receive calls over pstn with dahdi. Thanks. sean ___ -- 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] pstn hangs up: MWI no message waiting ??
Tilghman Lesher wrote: On Friday 16 January 2009 20:27:57 sean darcy wrote: Tilghman Lesher wrote: On Friday 16 January 2009 17:43:21 sean darcy wrote: Danny Nicholas wrote: Why not do a zap restart instead of restarting asterisk? You could write an AGI to do the ZR when the condition occurred and lines where empty. Yes, a cron job to restart zaptel would cut off any call then existing. But how would I test for it? I can imagine: exten=s,n,ExecIf(some damn thing, System(service dahdi restart)) It's the some damn thing I can't imagine. How do you test if dahdi is acting up? Not a service restart, but a dahdi restart. You can't restart the dahdi service without first stopping Asterisk, anyway. if [ `../asterisk-trunk/contrib/scripts/astcli core show channels | wc -l` = 3 ]; then asterisk -rx 'dahdi restart'; fi Wow. I'll try that tomorrow. Put it as the cmd right after answer(), right? Or maybe, h,1 ? Well anyway, at least I'll be able to receive calls over pstn with dahdi. No, I'd actually recommend that as a cron job. It's basically, restart if idle. Any possibility of actually fixing dahdi? sean ___ -- 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] pstn hangs up: MWI no message waiting ??
Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! As part of the implementation of issue 8587, a check was incldued for MWI messages preceded by Ring Pulse Alert Signals (RPAS). The RPAS is answered by chan_dahdi as a standard call and the MWI message is processed. As part of the implementation, if the MWI message was included, the channel was hung up. This did not take into account that possibility of MWI messages included into the to standard CID spills. I believe this is the case here and the MWI portion of the CID spill is causing the channel to hang up. You can look at commit 169154 for a fix or simply remove the ast_hangup calls immediately after the message MWI: channel %d no message waiting!\n and MWI: Channel %d no message w Thanks. That's a lot better idea than calling Digium Monday and yelling bloody murder. Why in the world would they screw up and obsolete their own hardware? sean ___ -- 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] pstn hangs up: MWI no message waiting ??
sean darcy wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! As part of the implementation of issue 8587, a check was incldued for MWI messages preceded by Ring Pulse Alert Signals (RPAS). The RPAS is answered by chan_dahdi as a standard call and the MWI message is processed. As part of the implementation, if the MWI message was included, the channel was hung up. This did not take into account that possibility of MWI messages included into the to standard CID spills. I believe this is the case here and the MWI portion of the CID spill is causing the channel to hang up. You can look at commit 169154 for a fix or simply remove the ast_hangup calls immediately after the message MWI: channel %d no message waiting!\n and MWI: Channel %d no message w Thanks. That's a lot better idea than calling Digium Monday and yelling bloody murder. Why in the world would they screw up and obsolete their own hardware? sean OK. Calmer now. If fact a 410 would have the same problem. I'll make the fix on our machines. Should I file a bug, or does the 169154 commit already fix it? sean ___ -- 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] pstn hangs up: MWI no message waiting ??
Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: Doug Bailey wrote: - sean darcy seandar...@gmail.com wrote: pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4 no message waiting! As part of the implementation of issue 8587, a check was incldued for MWI messages preceded by Ring Pulse Alert Signals (RPAS). The RPAS is answered by chan_dahdi as a standard call and the MWI message is processed. As part of the implementation, if the MWI message was included, the channel was hung up. This did not take into account that possibility of MWI messages included into the to standard CID spills. I believe this is the case here and the MWI portion of the CID spill is causing the channel to hang up. You can look at commit 169154 for a fix or simply remove the ast_hangup calls immediately after the message MWI: channel %d no message waiting!\n and MWI: Channel %d no message w Using 1.6.1-169788, now it just never answers: -- Starting simple switch on 'DAHDI/4-1' [Jan 22 15:33:01] NOTICE[28000]: chan_dahdi.c:7144 ss_thread: Got event 18 (Ring Begin)... [Jan 22 15:33:04] NOTICE[28000]: chan_dahdi.c:7144 ss_thread: Got event 2 (Ring/Answered)... [Jan 22 15:33:04] NOTICE[28000]: chan_dahdi.c:7316 ss_thread: MWI: Channel 4 no message waiting! and that's it. Started bug 14313. sean ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
Tilghman Lesher wrote:
On Tuesday 24 February 2009 13:44:25 Barry L. Kline wrote:
Here's one that may be of interest to any upgraders. If you rely on the
behavior of gosub you may want to make note of this change.
I have an incoming call context:
exten = _,n,GoSub(incoming,${EXTEN},1(${EXTEN}));
that is supposed to gosub into the incoming extension at priority 1.
Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the
requested extension wasn't present in the incoming context.
When I upgraded to 1.6.0.6 this behavior changed and I would simply get
an error on the console that a matching extension was not found, and the
dialplan would simply stop. It was easy enough to add:
[incoming]
exten = _,1,Goto(i,1)
to restore the previous behavior (I'm looking at four-digits from a PRI)
which I should probably have done anyway.
I don't know if this is a bug or WAD but just wanted to mention it.
It was a bug. Gosub/Goto should NEVER go to the i extension, unless that
target is explicitly given. The use of the i extension for invalid
extensions is limited to WaitExten/Background.
Why should it be so limited? It's clearly not now, and it's not been
considered a bug - certainly no bug reports or user confusion. Some of
us have used this behaviour for quite a while. It's very useful.
Why change?
sean
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Re: [asterisk-users] Gosub behavior change =1.6.0.5 to 1.6.0.6
Tilghman Lesher wrote: On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: Tilghman Lesher schrieb: On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: Barry L. Kline wrote: that is supposed to gosub into the incoming extension at priority 1. Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the requested extension wasn't present in the incoming context. Really strange that Goto and Gosub behave different. If Goto behaves that way, that's a bug. As stated in a prior email, the i extension should only be implicitly invoked when waiting for a new extension and the typed extension does not match anything. FYI: If you take a look at the history of http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension you will find out that the old behavior is there since at least Nov. 2005, and probably used since then. voip-info.org is best known for being often wrong. I think the point being made was that a lot of people thought this was a feature, not a bug. I assume you're asserting the the dev's did not expect this behaviour, even if a large group of users did. That's OK. But there's still the question about why this behaviour is so bad/inconsistent/something that it should be changed. Simply labeling it a bug is just a conclusion. Why is it a bug??? sean ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
Tilghman Lesher wrote: . ... but I absolutely defend fixing this bug in Gosub, given that I'm the designer of it, and it was never supposed to fail into the i extension. Wow. sean ___ -- 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] an easy way to deal with/without leading 1 ?
I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.
The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes that are toll free.
I tried
exten=exten = _!877NXX,2,Dial(${pstnline}/ww1${EXTEN:-10})
but that matches everything. I'd hoped it would only match strings
that had zero or more characters, followed by the 877 pattern.
sean
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[asterisk-users] an easy way to deal with/without leading 1 ?
I posted this before, but it didn't show up. So if it's a dup...
I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.
The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes that are toll free.
I tried
exten=exten = _!877NXX,2,Dial(${pstnline}/ww1${EXTEN:-10})
but that matches everything. I'd hoped it would only match strings
that had zero or more characters, followed by the 877 pattern.
sean
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Re: [asterisk-users] callpickup not working
On Sat, Mar 28, 2009 at 11:25 PM, Zvonimir Mileta zmil...@hotmail.com wrote: hi folks, Im pretty sure this has been covered before but I just wasnt able to find any answer. Im having troubles with the call pickup feature, is just not working for me. whenever I press *8 or 200 or anyother. nothing happens and sometimes I also get nothing to pickup. I have read this might be a bug although I havent found any patch for it. does anyone have any ideas? Im using BSD 7.1 with * 1.4.6 thanks in advance. --zvonimir ___ -- 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 call pickup didn't work in 1.4.23. It did in 1.4.22, and supposedly works in 1.4.24. But 1.4.6? That's a while ago! sean ___ -- 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] Anyone actually built h323plus on Fedora?
I've been trying to build h323plus (both the release and svn) for chan_h323 on Fedora 10. No joy. I posted on the h323plus ml, but no response. Anybody here actually built it on Fedora? Wanna share your secrets, or even better a specfile? sean ___ -- 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] [OFF TOPIC] wich virtualization solution to use?
On Sat, Apr 11, 2009 at 12:04 PM, David fire ddf...@gmail.com wrote: hi there are a lot of virtualization solution out there and every one is the best and has some pro and some cons... wich one do you recomend? the idea to isolate diferents servers asterisk apache ... it is a good idea? sorry for the off topic but here is a place where are a lot of linux gurus Thanks If all your virtual machines are linux, openvz is probably the easiest and provides the best performance. But all it does is linux. sean ___ -- 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] random hangups: how to debug?
I've a TDM400 with dahdi 2.1.0.4, asterisk 1.6.1-rc5. Asterisk is randomly hanging up calls coming over the pstn. Often it happens right as the call is answered: -- Starting simple switch on 'DAHDI/4-1' [Apr 22 17:09:38] NOTICE[20123]: chan_dahdi.c:7505 ss_thread: Got event 18 (Ring Begin)... [Apr 22 17:09:40] NOTICE[20123]: chan_dahdi.c:7505 ss_thread: Got event 2 (Ring/Answered)... -- Executing [...@incoming-pstn-line:1] Answer(DAHDI/4-1, ) in new stack -- Executing [...@incoming-pstn-line:2] Wait(DAHDI/4-1, 3) in new stack -- Executing [...@incoming-pstn-line:3] NoOp(DAHDI/4-1, callerid : ) in new stack -- Executing [...@incoming-pstn-line:4] Set(DAHDI/4-1, CALLERPRES()=allowed) in new stack -- Executing [...@incoming-pstn-line:5] Dial(DAHDI/4-1, DAHDI/1,60) in new stack -- Called 1 -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 answered DAHDI/4-1 -- Native bridging DAHDI/4-1 and DAHDI/1-1 -- Hungup 'DAHDI/1-1' == Spawn extension (incoming-pstn-line, s, 5) exited non-zero on 'DAHDI/4-1' -- Hungup 'DAHDI/4-1' but sometimes it happens after 2-3 minutes of conversation. I figure this is a dahdi problem, but how do I get a more detailed log of what's happening here? Is there a way of setting up debug messages only for dahdi? Since it's random, I'd be collecting a lot of junk to just set debug. sean ___ -- 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] 64bit: any problems with asterisk?
We're getting a new server. I'm considering installing 64bit fedora rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any issues we should expect? sean ___ -- 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] 64bit: any problems with asterisk?
John Novack wrote: Suggest you use CentOS rather than Fedora. CentOS has a longer support life, with the same cost. JMO John Novack sean darcy wrote: We're getting a new server. I'm considering installing 64bit fedora rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any issues we should expect? sean Thanks for all the responses. I didn't expect any issues with 64 bit, but... So I'm off to install this weekend. sean ___ -- 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] 1.6.1: menuselect has problems with x86_64 ??
1.6.1 svn 190575: CC=cc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent menuselect make[1]: Entering directory `/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect' gcc -m64 -march=native -mtune=native -floop-interchange -floop-strip-mine -floop-block -c -o menuselect_stub.o menuselect_stub.c gcc -o menuselect menuselect.o strcompat.o menuselect_stub.o mxml/libmxml.a /usr/bin/ld: i386 architecture of input file `menuselect.o' is incompatible with i386:x86-64 output /usr/bin/ld: i386 architecture of input file `strcompat.o' is incompatible with i386:x86-64 output sean ___ -- 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] 1.6.1: DNS error but ping works
With 1.6.1 svn: [2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout: -- Registration for '17470121...@proxy01.sipphone.com' timed out, trying again (Attempt #30) [2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable to lookup 'proxy01.sipphone.com' [2009-04-26 15:01:00] WARNING[1844]: chan_sip.c:10037 transmit_register: Probably a DNS error for registration to 1747yyyx...@proxy01.sipphone.com, trying REGISTER again (after 20 seconds) but ping works: ping proxy01.sipphone.com PING northamerica.sipphone.com (198.65.166.131) 56(84) bytes of data. 64 bytes from northamerica.sipphone.com (198.65.166.131): icmp_seq=1 ttl=52 time=96.5 ms 64 bytes from northamerica.sipphone.com (198.65.166.131): icmp_seq=2 ttl=52 time=94.4 ms Is this a bug, or could it be caused by a faulty configuration? sean ___ -- 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] 1.6.1 app_fax: WARNING T.30 ECM carrier not found ??
Receiving a fax with 1.6.1: == Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on 'DAHDI/4-1' -- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax Detected) in new stack -- Executing [...@incoming-pstn-line:2] Goto(DAHDI/4-1, incoming-fax,s,1) in new stack -- Goto (incoming-fax,s,1) -- Executing [...@incoming-fax:1] Set(DAHDI/4-1, FAXFILE=/var/spool/asterisk/fax/20090504_1602-0) in new stack -- Executing [...@incoming-fax:2] ReceiveFAX(DAHDI/4-1, /var/spool/asterisk/fax/20090504_1602-0.tif) in new stack -- Starting simple switch on 'DAHDI/1-1' -- Remote UNIX connection -- Hungup 'DAHDI/1-1' [2009-05-04 16:02:39] WARNING[12989]: app_fax.c:128 span_message: WARNING T.30 ECM carrier not found [2009-05-04 16:02:40] WARNING[12989]: app_fax.c:128 span_message: WARNING T.30 ECM carrier not found [2009-05-04 16:02:40] WARNING[12989]: app_fax.c:128 span_message: WARNING T.30 ECM carrier not found [2009-05-04 16:02:46] WARNING[12989]: app_fax.c:128 span_message: WARNING T.30 ECM carrier not found [2009-05-04 16:02:46] WARNING[12989]: app_fax.c:128 span_message: WARNING T.30 ECM carrier not found -- Executing [...@incoming-fax:3] Hangup(DAHDI/4-1, ) in new stack ECM - error correction mode ( right?) - but the fax is received OK. Any reason to worry? Anything to do? sean ___ -- 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] 1.6.1 app_fax: WARNING T.30 ECM carrier not found ??
David Backeberg wrote: On Mon, May 4, 2009 at 10:52 PM, sean darcy seandar...@gmail.com wrote: Receiving a fax with 1.6.1: == Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on 'DAHDI/4-1' -- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax Detected) in new stack -- Executing [...@incoming-pstn-line:2] Goto(DAHDI/4-1, incoming-fax,s,1) in new stack -- Goto (incoming-fax,s,1) -- Executing [...@incoming-fax:1] Set(DAHDI/4-1, FAXFILE=/var/spool/asterisk/fax/20090504_1602-0) in new stack -- Executing [...@incoming-fax:2] ReceiveFAX(DAHDI/4-1, /var/spool/asterisk/fax/20090504_1602-0.tif) in new stack -- Starting simple switch on 'DAHDI/1-1' -- Remote UNIX connection -- Hungup 'DAHDI/1-1' [2009-05-04 16:02:39] WARNING[12989]: app_fax.c:128 span_message: WARNING T.30 ECM carrier not found [2009-05-04 16:02:40] WARNING[12989]: app_fax.c:128 span_message: WARNING T.30 ECM carrier not found [2009-05-04 16:02:40] WARNING[12989]: app_fax.c:128 span_message: WARNING T.30 ECM carrier not found [2009-05-04 16:02:46] WARNING[12989]: app_fax.c:128 span_message: WARNING T.30 ECM carrier not found [2009-05-04 16:02:46] WARNING[12989]: app_fax.c:128 span_message: WARNING T.30 ECM carrier not found -- Executing [...@incoming-fax:3] Hangup(DAHDI/4-1, ) in new stack ECM - error correction mode ( right?) - but the fax is received OK. Any reason to worry? Anything to do? Since you are receiving a fax over an analog line, you can set up MixMonitor() or Monitor() to record the fax, play it back, listen for line noise or static or something else that may be happening that's throwing those warnings from SpanDSP. Obviously if you're getting a fax you're not as worried as if this caused the fax to drop. Once you have a recording of the transmission and the logs of what happens you can submit it to the SpanDSP project to see what they think. Incidentally, how verbose do you have to set your CLI to get output that lists the warnings? Just asterisk -r It shows: Verbosity is at least 3 sean ___ -- 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] how to avoid call waiting? Or check DIALSTATUS before Dial()?
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =s,1,Answer()
exten =s,n,Dial(${mainline},60)
exten =s,n,ExecIf($[${DIALSTATUS} = BUSY]?Dial(${secondline},30))
But it doesn't work because * first tries Call Waiting on the main line.
Here I dial out:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@internal:2] Set(DAHDI/1-1, CALLERID=house
2127873453) in new stack
-- Executing [...@internal:3] Dial(DAHDI/1-1,
.
And now an incoming call:
-- Executing [...@incoming:1] Answer(IAX2/nhi-10929, ) in new stack
-- Executing [...@incoming:2] Dial(IAX2/nhi-10929, DAHDI/1,60) in
new stack
-- Called 1
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- DAHDI/1-2 is ringing
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- CPE supports Call Waiting Caller*ID. Sending 'Seandarcy/212 573
1432'
Is there a way to check the status of a dahdi channel _before_ dialing it?
exten =s,n,ExecIf($[DAHDI/1${DIALSTATUS} =
BUSY]?Dial(${secondline},30)) ??
What's special control 20 ??
Any help appreciated.
sean
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Re: [asterisk-users] how to avoid call waiting? Or check DIALSTATUS before Dial()?
sean darcy wrote:
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =s,1,Answer()
exten =s,n,Dial(${mainline},60)
exten =s,n,ExecIf($[${DIALSTATUS} = BUSY]?Dial(${secondline},30))
But it doesn't work because * first tries Call Waiting on the main line.
Here I dial out:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@internal:2] Set(DAHDI/1-1, CALLERID=house
2127873453) in new stack
-- Executing [...@internal:3] Dial(DAHDI/1-1,
.
And now an incoming call:
-- Executing [...@incoming:1] Answer(IAX2/nhi-10929, ) in new stack
-- Executing [...@incoming:2] Dial(IAX2/nhi-10929, DAHDI/1,60) in
new stack
-- Called 1
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- DAHDI/1-2 is ringing
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- CPE supports Call Waiting Caller*ID. Sending 'Seandarcy/212 573
1432'
Is there a way to check the status of a dahdi channel _before_ dialing it?
exten =s,n,ExecIf($[DAHDI/1${DIALSTATUS} =
BUSY]?Dial(${secondline},30)) ??
What's special control 20 ??
Any help appreciated.
sean
BTW, this is on 1.6.1.
sean
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[asterisk-users] howto build oslec with dahdi-linux-2.1.0.4 or svn?
On Fedora 11, gcc-4.4, I'm trying to build oslec in dahdi-linux, but: [aster...@asterisk dahdi-linux]$ make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/firmware' make[1]: Leaving directory `/home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/firmware' make -C /lib/modules/2.6.29.3-140.fc11.x86_64/build SUBDIRS=/home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi DAHDI_INCLUDE=/home/asterisk/build/dahdi/svn/dahdi-linux/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[1]: Entering directory `/usr/src/kernels/2.6.29.3-140.fc11.x86_64' CC [M] /home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.o /home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.c: In function ‘echo_can_free’: /home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.c:70: error: implicit declaration of function ‘oslec_free’ /home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.c: In function ‘echo_can_process’: /home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.c:82: error: implicit declaration of function ‘oslec_update’ /home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.c: In function ‘echo_can_create’: /home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.c:103: error: implicit declaration of function ‘oslec_create’ Also tried dahdi svn - same result. sean ___ -- 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 to avoid call waiting? Or check DIALSTATUS before Dial()?
Rilawich Ango wrote:
Can you try to disable call waiting in your phone?
On Fri, May 15, 2009 at 6:44 AM, sean darcy seandar...@gmail.com wrote:
sean darcy wrote:
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =s,1,Answer()
exten =s,n,Dial(${mainline},60)
exten =s,n,ExecIf($[${DIALSTATUS} = BUSY]?Dial(${secondline},30))
But it doesn't work because * first tries Call Waiting on the main line.
Here I dial out:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@internal:2] Set(DAHDI/1-1, CALLERID=house
2127873453) in new stack
-- Executing [...@internal:3] Dial(DAHDI/1-1,
.
And now an incoming call:
-- Executing [...@incoming:1] Answer(IAX2/nhi-10929, ) in new stack
-- Executing [...@incoming:2] Dial(IAX2/nhi-10929, DAHDI/1,60) in
new stack
-- Called 1
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- DAHDI/1-2 is ringing
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- IAX2/nhi-10929 requested special control 20, passing it to DAHDI/1-2
-- CPE supports Call Waiting Caller*ID. Sending 'Seandarcy/212 573
1432'
Is there a way to check the status of a dahdi channel _before_ dialing it?
exten =s,n,ExecIf($[DAHDI/1${DIALSTATUS} =
BUSY]?Dial(${secondline},30)) ??
What's special control 20 ??
Any help appreciated.
sean
BTW, this is on 1.6.1.
sean
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How? It's only an analogue extension.
sean
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Re: [asterisk-users] how to avoid call waiting? Or check DIALSTATUS before Dial()?
Tzafrir Cohen wrote:
On Thu, May 14, 2009 at 06:37:53PM -0400, sean darcy wrote:
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =s,1,Answer()
exten =s,n,Dial(${mainline},60)
exten =s,n,Dial(DAHDI/g5,60)
For both channels set: group = 5
So in chan_dahdi.conf:
context=internal ; Uses the [internal] context in extensions.conf
signalling=fxo_ks ; fxo_ks not auto Use FXO signalling for an FXS
channel - as set in sytem.conf.conf
group = 5
channel = 1 ; Telephone attached to port 1
channel = 2 ; Telephone attached to port 2
;;dahdichan = 1,2
exten =s,n,ExecIf($[${DIALSTATUS} = BUSY]?Dial(${secondline},30))
But it doesn't work because * first tries Call Waiting on the main line.
Disabling that can also help.
How do I disable call waiting on dahdi 1 and 2, the internal extensions
- which are simple POTS phones off the TDM400P - but leave it on for
pstn-in?
Trying to answer my own question:
Put callwaiting = no in chan_dahdi.conf,
then the internal stanza above,
then callwaiting = yes,
followed by the pstn stanza?
Thanks for the help.
sean
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Re: [asterisk-users] how to avoid call waiting? Or check DIALSTATUS before Dial()?
Yehavi Bourvine wrote: You check for BUSY. Check for IN_USE instead. That's what I do here (on 1.4, but I guess that 1.6 behaves similarly). When an extension is in IN_USE state I have a decision tree after consulting a database: * If the user wants waiting call - dial him/her/ * If the user doesn;t want waiting call but wants voicemail answer - send to voicemail with B prefix. * If niether is wanted - play busy. Regards, __Yehavi: Great. Do you have a dialplan snippet you'd be willing to share? sean ___ -- 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
