[Asterisk-Users] strange problem with SIP/voicemail
I'm having a very strange problem I've been fighting with all day. It's 2:30am, and I'm stuck. I think the problem may lie with one of my SIP providers, but I'm not sure. I have two ways to call into my test Grandstream. I can call a PSTN 360 area code number that will forward to my FWD number, which in turn is registered with my * box on extension 2030. If I call the 360 number, everything works, my Grandstream rings, and if I don't answer, it goes to voicemail and voicemail works. I also have a PSTN 972 area code number that forwards directly to my * box. If I call the 972 number, my Grandstream will ring, but if I don't answer, it will give me silence for a bit, then I hear a click, my CLI interface says that it is recording a message, but then it says: Apr 19 02:21:20 WARNING[15373]: app_voicemail.c:1261 play_and_record: No audio available on SIP/66.147.170.34-0811abe8?? Here is my exten map [actual phone number munged]. I have removed the Grandstream from the exten for this example. It makes no difference whether the Grandstream gets rang or not: exten = 9725551212,1,Answer exten = 9725551212,2,Voicemail2(u1000) exten = 9725551212,3,Hangup Also, just for testing, I have added this extension: exten = 2501,1,Voicemail2(u1000) exten = 2501,2,Hangup If I dial 2501 from my grandstream, voicemail works that way, too. My questions: 1) Should I have the Answer in there or not? It doesn't help to add or remove it. On the FWD number, I do not have an Answer. 2) I can get voicemail to work on the incoming 972 number if I change the dialplan around and then do a restart gracefully. Example: exten = 9725551212,1,Answer exten = 9725551212,2,Playback(transfer) exten = 9725551212,3,Voicemail2(u1000) exten = 9725551212,4,Hangup It will work once, maybe twice, and then it won't work any more after that until I fiddle with the dialplan again and do another restart. On Saturday when I thought I had all of this working, I dialed in at least ten times and had no problems. I originally was running a CVS from 03-14-04 now I am running 04-19-04, and still have the same issue. Anyone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Matching variable-length extensions with chan_zap in overlap dialling
Jeremy McNamara wrote:
Try exten = _0X. --- notice the period
[m807oth]
exten = _80780780.,1,StripMSD(7)
exten = _0.,1,SetVar,clidest=${EXTEN}
exten = _0.,2,Goto(cli,s,1)
...noticed mine? :-) I've tried a combo-wildcard (with an X, as in your
example) as well, with no results either. The code in chan_zap.c seems
to confirm that in overlap digit transmission the channel driver doesn't
check for multiple matches. The patch to check for
multiple/ambiguous/possibly incomplete matches is trivial, but
implementing the timeout is definitely not.
Apollon Koutlides
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Re: [Asterisk-Users] Does RTP traffic go through Asterisk IP PBX ?
PTCHEN wrote: Is there anybody knows if RTP traffic goes thru Asterisk IP PBX? If it is, it must limit the capacity of Asterisk. Do you know the concurrent SIP call capacity? And Is there any guy modify the source code to prevent this? Can be done already: http://voip-info.org/wiki-Asterisk+Letting+SIP+clients+connect+directly F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI Eicon Diva Server 4BRI
Hi all, I have a PC working with a DIVA Eicon Server 4BRI during a lot of time. Now I can't make call but I can receive calls. I load diva with command: divactrl load -c 1 -f ETSI -u -t 0 Country: Spain Isdnmode: point to point My capi.conf is the next: [global] mode=immediate isdnmode=ptp txgain=0.8 rxgain=0.5 [interfaces] msn=952901652,952901987 incomingmsn=* controller=1,2 softdtmf=0 context=default echocancel=1 echotail=64 callgroup=1 devices=4 I obtnain next trace in console: -- Attempting call on CAPI/@952901987:B951014947||r for [EMAIL PROTECTED]:2 (Retry 1) -- data = @952901987:B951014947||r -- capi request omsn = @952901987 == found capi with omsn = 952901987 == CAPI Call CAPI[contr1/952901987]/7 with B3 == CAPI Call CAPI[contr1/952901987]/7 with B3-- creating pipe for PLCI=-1 -- CONNECT_CONF ID=001 #0x0f52 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 == received CONNECT_CONF PLCI = 0x301 INFO = 0 == DISCONNECT_IND PLCI=0x301 REASON=0x3302 -- CAPI Hangingup -- removed pipe for PLCI = 0x301 Apr 19 12:03:17 NOTICE[25619]: pbx_spool.c:199 attempt_thread: Call failed to go through, reason 1 Any idea? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange CallerId behaviour with SIP
Hi all,
I want to see the name of the caller (if available) and not the number.
If I call from my IP phone to my software IP phone I see the name of the caller. If I
call from the software phone to my IP phone I only see the number, not the name. If I
call from IP phone to IP phone I only see the number. If I set the name explicitely
using callerid = asrevieved in my sip .conf or if I use SetCallerID(${CALLERIDNAME})
I get aseterisk on my display. IAX calls seem to go OK.
Is this known behaviour or is my configuration wrong? If so, any hints for a sollution?
Groeten,
Joost Kraaijeveld
Askesis B.V.
Molukkenstraat 14
6524NB Nijmegen
tel: 024-3888063 / 06-51855277
fax: 024-3608416
e-mail: [EMAIL PROTECTED]
web: www.askesis.nl
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RE: [Asterisk-Users] CAPI Eicon Diva Server 4BRI
Hi, Executing divactrl dchannel -dmonitor -Debug I obtain the next messages: MDL-ERROR(G) SIG-EVENT 0A MDL-ERROR(G) SIG-EVENT 0A MORE SIG-X(045) 08 01 12 05 A1 04 03 80 90 A3 18 01 81 6C 0B 00 A0 39 35 32 39 30 31 39 38 37 70 0D 80 39 35 31 30 31 34 39 34 37 7C 7C 72 7D 02 91 81 Q.931 CR12 SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 81 Calling Party Number 00 a0 '952901987' Called Party Number 80 '951014947||r' HLC 91 81 MDL-ERROR(G) SIG-EVENT 0A SIG-EVENT 0A EVENT: Call failed in State 'Call initiated' Link disconnected, TEI error MDL-ERROR(G) SIG-EVENT 0A MDL-ERROR(G) SIG-EVENT 0A MDL-ERROR(G) SIG-EVENT 0A Any idea? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Revuelto Enviado el: miércoles, 19 de mayo de 2004 12:00 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] CAPI Eicon Diva Server 4BRI Hi all, I have a PC working with a DIVA Eicon Server 4BRI during a lot of time. Now I can't make call but I can receive calls. I load diva with command: divactrl load -c 1 -f ETSI -u -t 0 Country: Spain Isdnmode: point to point My capi.conf is the next: [global] mode=immediate isdnmode=ptp txgain=0.8 rxgain=0.5 [interfaces] msn=952901652,952901987 incomingmsn=* controller=1,2 softdtmf=0 context=default echocancel=1 echotail=64 callgroup=1 devices=4 I obtnain next trace in console: -- Attempting call on CAPI/@952901987:B951014947||r for [EMAIL PROTECTED]:2 (Retry 1) -- data = @952901987:B951014947||r -- capi request omsn = @952901987 == found capi with omsn = 952901987 == CAPI Call CAPI[contr1/952901987]/7 with B3 == CAPI Call CAPI[contr1/952901987]/7 with B3-- creating pipe for PLCI=-1 -- CONNECT_CONF ID=001 #0x0f52 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 == received CONNECT_CONF PLCI = 0x301 INFO = 0 == DISCONNECT_IND PLCI=0x301 REASON=0x3302 -- CAPI Hangingup -- removed pipe for PLCI = 0x301 Apr 19 12:03:17 NOTICE[25619]: pbx_spool.c:199 attempt_thread: Call failed to go through, reason 1 Any idea? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Accommodating multiple FWD users
On Sun, 18 Apr 2004, Malcolm Taylor wrote:
Can anyone suggest a way in which all users could dial the prefix 8 and *
would automatically associate the correct FWD account for the outbound call?
I had to do something like this for outgoing calls over two different PSTN
lines. It's probably sub-optimal, but at least I did not have to do
anything with database or other code...
extensions.conf: (watch for split / wrap on the last exten line)
[globals]
TRUNKCOMPANY=Zap/1
TRUNKCUSTOMER=Zap/1
TRUNKHOMELINE=Zap/2
PREFIXCOMPANY=
PREFIXCUSTOMER=*11*2#W
PREFIXHOMELINE=
[macro-trunkdial]
exten = s,1,Dial(${ARG2}/${ARG3}${ARG1})
exten = s,2,Congestion
[homeline]
include = allnumbers
[company]
include = allnumbers
[customer]
include = allnumbers
[allnumbers]
exten =
_9.,1,Macro(trunkdial|${EXTEN:1}|${TRUNK${CONTEXT}}|${PREFIX${CONTEXT}})
In the technology configuration files (zapata.conf, sip.conf, etc) I put
the user into the relevant context for the line I want them to dial out
on. (The ${PREFIX...} variable is used to control an override dialling
code, for example to select a long-distance provider or--in my case--to
charge calls to a second number on the line.)
I could probably make the macro a bit simpler by using ${MACROEXTEN},
${MACROCONTEXT}, etc instead of passing three arguments. Maybe I'll look
at that for version two.
Anyway, this saved me from duplicating all the outgoing definitions in my
dialplan (it is a lot more complicated than the _9. I show above). If
you've already got different extens set up for the various FWD accounts,
this method should not be too hard to adapt for your purposes.
Hope it helps,
Vic Cross
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Re: [Asterisk-Users] Accommodating multiple FWD users
On Sun, 2004-04-18 at 20:37, Malcolm Taylor wrote:
I have five SIP users on my * box, each of whom has his own FWD account.
Right now I have my configuration set so that the first user dials the
prefix 8 when calling to an FWD number, the second user dials the prefix 7
and so on. This way, the FWD user he is calling sees the correct Caller ID
information.
Can anyone suggest a way in which all users could dial the prefix 8 and *
would automatically associate the correct FWD account for the outbound call?
Try using GoToIf [show application gotoif] in combination with
${CALLERIDNUM} [asterisk/doc/README.variables]
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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Re: [Asterisk-Users] Strange CallerId behaviour with SIP
Joost Kraaijeveld wrote:
Hi all,
I want to see the name of the caller (if available) and not the number.
If I call from my IP phone to my software IP phone I see the name of the caller. If I call from the software phone to my IP phone I only see the number, not the name. If I call from IP phone to IP phone I only see the number. If I set the name explicitely using callerid = asrevieved in my sip .conf or if I use SetCallerID(${CALLERIDNAME}) I get aseterisk on my display. IAX calls seem to go OK.
Is this known behaviour or is my configuration wrong? If so, any hints for a sollution?
Could be the phone, some phones dont do Alpha characters, the
Grandstream for example can not display the name, only the number.
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Re: [Asterisk-Users] Intel 536ep as a FXO?
As someone who used to adjust hybrids for a living a number of years ago, I can tell you, complex impedence matching is only a part of the equation. Same here. The most important part is proper gain structure. If that's wrong no there is no way to control echo. No amount of tweaking of compensation networks will bring one into balance... No Convolution processing can control it. On old style equipment i.e. stuff built by Tellabs, the gain structure had to be right within about .5 DBm0. Alignment meant dialing up a milliwatt test signal, measuring that signal at the 2 wire point and adjusting pads on the module so that the 4 wire transmit point was at a fixed and correct level. If memory serves, on an analog microwave system, 0 DBm into a module was supposed to be -16 DBm on the 4 wire transmit point. The picture below may help to clarify: A major part of the problem implementing * into a pstn environment is that few implementors actually understand transmission basics, a smaller percentage actually have the test gear to measure the values, and even a smaller number understand what impedence, DBm, noise levels, twisted pair, induction, etc, mean in terms of pstn interface performance. Combine that with dropping an FXO interface into a pstn environment where the transmission levels to the CO are basically unknown, SOHO impedence mismatches abound, bridged analog phone sets are commonplace, and assumptions that plug-n-play applies across the board including the x100p, its fairly obvious why so many people bad-mouth the hardware. Its also interesting that in about eight months on this list no one has asked what the milliwatt generator is for, how to find the telephone number of the pstn generator, how to measure the levels or what the objectives should be, etc. The transmission levels that were noted in the original posting are those associated with the analog toll network, but the principle still applies. Maybe a couple of us should write a whitepaper for beginners on the topic. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP dropouts
Howdy all... When making SIP calls through my X100P from X-Lite to the PSTN I'm getting 3-5 second dropouts in both directions. I've tried ulaw and GSM, but that doesn't seem to make a difference, and the * box is on my local net. Here's my hardware: Celeron 2.4GHz, 512MB, Slackware 9.1, 2 X100P, 1 T100P. Any ideas what could be happening, or pointers as to how to shoot this trouble? Thanks, Brad Waite ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intel 536ep as a FXO?
On Mon, 19 Apr 2004, Rich Adamson wrote: Maybe a couple of us should write a whitepaper for beginners on the topic. Yes Please do Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intel 536ep as a FXO?
Combine that with dropping an FXO interface into a pstn environment where the transmission levels to the CO are basically unknown, SOHO impedence mismatches abound, bridged analog phone sets are commonplace, and assumptions that plug-n-play applies across the board including the x100p, its fairly obvious why so many people bad-mouth the hardware. I am badmouthing the hardware because I can drop an Adit600 FXO port on to the exact same line and have an order of magnitude better chance of getting adequate voice quality out of it. I am waiting for my FXO module to arrive so I can see if I have similar experiences with it. The X100P is a cheap hybrid interface. I am not arguing that point. I also believe, however, that using the X100P and reselling that particular brand of WinModem is giving a *lot* of asterisk newcomers a very bad taste in their mouths. It is my sincere hope that the TDM400P's FXO module is a significantly better hybrid and that the Dev Kit is simply a TDM400P with FXS and FXO modules. There's always a tradeoff between cost and performance. It is my opinion that the X100P was a bad choice. Its also interesting that in about eight months on this list no one has asked what the milliwatt generator is for, how to find the telephone number of the pstn generator, how to measure the levels or what the objectives should be, etc. I am pretty sure that most people wouldn't have the means to measure and apply that knowlege. I know what a milliwatt generator is used for and I have the means to measure and adjust the hybrid to get the desired result, but I didn't have the knowledge that yourself and Mr. Adamson have just brought to the list. In other words, I didn't know _where_ I needed to adjust the values to. It's especially interesting how the hybrid should NOT be adjusted to get 0dBm on the 4-wire side to eliminate echo. I would not have guessed that. I was also lucky enough not to need to live with the X100P for very long. Maybe a couple of us should write a whitepaper for beginners on the topic. I think that would be an incredible nugget of knowledge for the Asterisk community. I know that I've got yours and Mr. Ferrell's messages stored away in my knowledgebase. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP dropouts
Sorry, forgot to mention that I set up an extension to play back a long MP3. Other than the occasional 20ms packet being dropped on the floor, no other detectable dropouts. Again, thanks for any pointers. Brad Waite ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speaking digits and time...
-- Executing DateTime(SIP/phone1-07ff, ) in new stack -- Playing '/var/lib/asterisk/sounds/digits/day-1' (language 'en') -- Playing '/var/lib/asterisk/sounds/digits/mon-3' (language 'en') -- Playing '/var/lib/asterisk/sounds/digits/h-19' (language 'en') This works - the pathname is complete - Joy. -- Executing SayDigits(SIP/phone1-0e7d, 203) in new stack -- Playing 'digits/2' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/3' (language 'en') This doesn't (silence). Path looks incomplete. Where in the source do I fix this -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Accommodating multiple FWD users
Hi!
Can anyone suggest a way in which all users could dial the prefix 8 and *
would automatically associate the correct FWD account for the outbound call?
Try using GoToIf [show application gotoif] in combination with
${CALLERIDNUM} [asterisk/doc/README.variables]
I prefer a slightly cleaner method:
1. create a type=peer entry for each outgoing FWD account in sip.conf.
For example you have [fwd-out_joe], [fwd-out_bob], [fwd-out_mary] where
you specify the individual username, fromuser and password
2. in sip.conf put each of your local phone users into their own context
like context=from-joe or context=from-mary
3. in extension.conf you do smth like include = default for each of
those person contexts like [from-joe], and arrange a FWD dialout like
_8X. = Dial(SIP/{EXTEN:[EMAIL PROTECTED])
If, however, you want to avoid individual contexts in extensions.conf you
could instead us DBput() and DBget() lookup the correct fwd-out_xxx
string based upon your local users's callerid.
Cheers, Philipp
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[Asterisk-Users] Advanced queueing
Hullo :) Please be gentle with me, I don't have a working * install, and am just looking for background information. I'm always impressed by companies who implement a queue like You are now number N in the queue. There are currently M agents answering calls, and your call should be answered in approx. O minutes I've seen on http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+additional that Allison has recorded soundfiles to support this style of queue, but how do I make use of them in Asterisk? Is there a pre-written application to implement this type of queue, or would it need to be an AGI-based affair? Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
I for one would love this. I do not have any test equipment to determine the level I am sending at, but if I could at least figure out what levels to have my rxgain values set to, that would help. I remember seeing somewhere that you can use a program (part of the zt suite if I remember correctly) to view the audio levels on the FXO card like an on-screen vu meter. I can use that and dial up my telco milliwatt test number and adjust accordingly. I asked where that tool was on the IRC channel, but they seemed to not know either. I have searched as I know I saw it, but can't find it again. Please post a guide like this to the Wiki or some other location, and be assured it will help at least one person out, probably many more. -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Monday, April 19, 2004 6:43 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Intel 536ep as a FXO? As someone who used to adjust hybrids for a living a number of years ago, I can tell you, complex impedence matching is only a part of the equation. Same here. The most important part is proper gain structure. If that's wrong no there is no way to control echo. No amount of tweaking of compensation networks will bring one into balance... No Convolution processing can control it. On old style equipment i.e. stuff built by Tellabs, the gain structure had to be right within about .5 DBm0. Alignment meant dialing up a milliwatt test signal, measuring that signal at the 2 wire point and adjusting pads on the module so that the 4 wire transmit point was at a fixed and correct level. If memory serves, on an analog microwave system, 0 DBm into a module was supposed to be -16 DBm on the 4 wire transmit point. The picture below may help to clarify: A major part of the problem implementing * into a pstn environment is that few implementors actually understand transmission basics, a smaller percentage actually have the test gear to measure the values, and even a smaller number understand what impedence, DBm, noise levels, twisted pair, induction, etc, mean in terms of pstn interface performance. Combine that with dropping an FXO interface into a pstn environment where the transmission levels to the CO are basically unknown, SOHO impedence mismatches abound, bridged analog phone sets are commonplace, and assumptions that plug-n-play applies across the board including the x100p, its fairly obvious why so many people bad-mouth the hardware. Its also interesting that in about eight months on this list no one has asked what the milliwatt generator is for, how to find the telephone number of the pstn generator, how to measure the levels or what the objectives should be, etc. The transmission levels that were noted in the original posting are those associated with the analog toll network, but the principle still applies. Maybe a couple of us should write a whitepaper for beginners on the topic. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Advanced queueing
Title: RE: [Asterisk-Users] Advanced queueing Position and hold time announcements/settings are in queues.conf in the later cvs versions. Matt -Original Message- From: Gavin Hamill [mailto:[EMAIL PROTECTED]] Sent: Monday, April 19, 2004 10:48 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Advanced queueing Hullo :) Please be gentle with me, I don't have a working * install, and am just looking for background information. I'm always impressed by companies who implement a queue like You are now number N in the queue. There are currently M agents answering calls, and your call should be answered in approx. O minutes I've seen on http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+additional that Allison has recorded soundfiles to support this style of queue, but how do I make use of them in Asterisk? Is there a pre-written application to implement this type of queue, or would it need to be an AGI-based affair? Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
I for one would love this. I do not have any test equipment to determine the level I am sending at, but if I could at least figure out what levels to have my rxgain values set to, that would help. I remember seeing somewhere that you can use a program (part of the zt suite if I remember correctly) to view the audio levels on the FXO card like an on-screen vu meter. I can use that and dial up my telco milliwatt test number and adjust accordingly. I asked where that tool was on the IRC channel, but they seemed to not know either. I have searched as I know I saw it, but can't find it again. The tool you're looking for is /usr/src/zaptel/ztmonitor [EMAIL PROTECTED] zaptel]# ./ztmonitor Usage: ztmonitor channel num [-v] [-f FILE] [EMAIL PROTECTED] zaptel]# ./ztmonitor 1 -v Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ##* Keep in mind that tool is nothing more then an audio VU meter and was not intended to be an accurate means of measuring transmission levels. I think bkw (probably with Mark) wrote it back in the November/December timeframe as a simple tool for adjusting rxgain, etc. About that same time, the echo cancelling mechanism (for the x100p) was rewritten to sense the audio reflection (or echo) during the first half-second or so of an initial pstn call. (That was a substantial improvement over previous cancellation methods without a doubt. If I recall recorrectly, that mechanism was reduced to sending an outbound short duration pulse or burst, and measuring the reflected energy. Sort of a snapshot at the start of an analog call. It's okay, but certainly not the equivalent of commercial analog cancellation products including mux's.) I've not had to revisit the x100p gain adjustment effort for several months, but seems to me that it was necessary to completely stop and start * each time an adjustment was made to the rxgain/txgain settings in zapata.conf (a simple reload wasn't adequate). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advanced queueing
On Monday 19 April 2004 16:13, Matthew Branton wrote: Position and hold time announcements/settings are in queues.conf in the later cvs versions. Superb - thanks for the speedy response :) Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk database support (SIP_FRIENDS)
On Sat, Apr 17, 2004 at 05:36:06PM +0200, Brancaleoni Matteo wrote: simply, you can define sip friends from a database. just create the table, enable SIP_FRIENDS into channels USE_SIP_MYSQL_FRIENDS=1 Makefile and read chan_sip.c how to set db access (db access data must be into sip.conf) dbname= ; Name of database dbhost=localhost; Hostname of server dbuser= ; MySQL user name dbpass= ; Password for dbuser ./contrib/scripts/sip-friends.sql can be used to create a sipfriends table in the database. Select and update priviledges for the dbuser on the sipfriends table are required. Insert (name,secret,context) records into sipfriends. Default values will be used for the other fields until your sip clients register. The secret is plaintext. But using MYSQL_FRIENDS is not just a replacement for a list of configuration data in sip.conf or some file included from there. CLI command sip show peers will not show anything and it is not possible to specify additional options. dtmfmode=rfc2833 default should work, account information may not be used, but there is no mailbox notification to the phone. In the latter case it is not an option which could be easily added by some minor modification in chan_sip.c. If you need additonal options, look at retrieve_sip_conf_from_mysql.pl It can be used to generate sip configuration from MySQL data. -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange CallerId behaviour with SIP
Could be the phone, some phones dont do Alpha characters, the
Grandstream for example can not display the name, only the number.
No, IAX calls go OK. Also as mentioned in my previous mail, if I set the name
explicitely using callerid = asrevieved in my sip .conf or if I use
SetCallerID(${CALLERIDNAME}) I get asterisk on my display. So I assume that the
phone can do Aplha characters.
Groeten,
Joost Kraaijeveld
Askesis B.V.
Molukkenstraat 14
6524NB Nijmegen
tel: 024-3888063 / 06-51855277
fax: 024-3608416
e-mail: [EMAIL PROTECTED]
web: www.askesis.nl
___
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RE: [Asterisk-Users] Intel 536ep as a FXO?
The next question for me is: How do I found out my telco milliwatt test number? I'm in Washington State using Qwest. The way I understand this, I'm to dialup the telco milliwatt test number and adjust the rxgain values using ztmonitor tool until the Max Audio Hit is in the middle of the bar graph for a normal conversation? Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, April 19, 2004 9:01 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? I for one would love this. I do not have any test equipment to determine the level I am sending at, but if I could at least figure out what levels to have my rxgain values set to, that would help. I remember seeing somewhere that you can use a program (part of the zt suite if I remember correctly) to view the audio levels on the FXO card like an on-screen vu meter. I can use that and dial up my telco milliwatt test number and adjust accordingly. I asked where that tool was on the IRC channel, but they seemed to not know either. I have searched as I know I saw it, but can't find it again. The tool you're looking for is /usr/src/zaptel/ztmonitor [EMAIL PROTECTED] zaptel]# ./ztmonitor Usage: ztmonitor channel num [-v] [-f FILE] [EMAIL PROTECTED] zaptel]# ./ztmonitor 1 -v Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ##* Keep in mind that tool is nothing more then an audio VU meter and was not intended to be an accurate means of measuring transmission levels. I think bkw (probably with Mark) wrote it back in the November/December timeframe as a simple tool for adjusting rxgain, etc. About that same time, the echo cancelling mechanism (for the x100p) was rewritten to sense the audio reflection (or echo) during the first half-second or so of an initial pstn call. (That was a substantial improvement over previous cancellation methods without a doubt. If I recall recorrectly, that mechanism was reduced to sending an outbound short duration pulse or burst, and measuring the reflected energy. Sort of a snapshot at the start of an analog call. It's okay, but certainly not the equivalent of commercial analog cancellation products including mux's.) I've not had to revisit the x100p gain adjustment effort for several months, but seems to me that it was necessary to completely stop and start * each time an adjustment was made to the rxgain/txgain settings in zapata.conf (a simple reload wasn't adequate). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
Hi Am Mo, 2004-04-19 um 16.50 schrieb Jeremy Hall: I remember seeing somewhere that you can use a program (part of the zt suite if I remember correctly) to view the audio levels on the FXO card like an on-screen vu meter. I can use that and dial up my telco milliwatt test number and adjust accordingly. I asked where that tool was on the IRC channel, but they seemed to not know either. I have searched as I know I saw it, but can't find it again. That would be ztmonitor, i guess: silverbox:/usr/src/build/rc20/zaptel # ./ztmonitor 2 -v Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spanish translation
Hi! Anyone know if there are a spanish translation? Thanks! Jorge de Jesus Ramirez Sanchez Calle Jirafa # 3903. Col. Lomas del Sol C.P. 31167. Chihuahua, Chih. México. Tel: +52 (614) 498-7223 Fax: +52 (614) 421-2306 Cel: +52 (614) 345-9098 url: http://kokey.gluch.org.mx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
On Mon, 19 Apr 2004, Jeremy Hall wrote: This may not be the case in all areas, but in my area with Qwest as well, all exchanges have the test at xxx-9996. For example, my number is in the 208 area code, 459 exchange, so the full number would be 208-459-9996. It is not tied to any specific number, so I can use any exchange local to me such as 323-9996. It may or may not work in your area, so try not to do it at 3:00 AM until you have verified the number. I'm also in a Qwest area, but that number doesn't work here. All of the techs that I have asked gave it to me with no problems. They are shy about the automatic ANI number, however... dave -Original Message- From: Ed Rubright [mailto:[EMAIL PROTECTED] Sent: Monday, April 19, 2004 9:51 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? The next question for me is: How do I found out my telco milliwatt test number? I'm in Washington State using Qwest. The way I understand this, I'm to dialup the telco milliwatt test number and adjust the rxgain values using ztmonitor tool until the Max Audio Hit is in the middle of the bar graph for a normal conversation? Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, April 19, 2004 9:01 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? I for one would love this. I do not have any test equipment to determine the level I am sending at, but if I could at least figure out what levels to have my rxgain values set to, that would help. I remember seeing somewhere that you can use a program (part of the zt suite if I remember correctly) to view the audio levels on the FXO card like an on-screen vu meter. I can use that and dial up my telco milliwatt test number and adjust accordingly. I asked where that tool was on the IRC channel, but they seemed to not know either. I have searched as I know I saw it, but can't find it again. The tool you're looking for is /usr/src/zaptel/ztmonitor [EMAIL PROTECTED] zaptel]# ./ztmonitor Usage: ztmonitor channel num [-v] [-f FILE] [EMAIL PROTECTED] zaptel]# ./ztmonitor 1 -v Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ##* Keep in mind that tool is nothing more then an audio VU meter and was not intended to be an accurate means of measuring transmission levels. I think bkw (probably with Mark) wrote it back in the November/December timeframe as a simple tool for adjusting rxgain, etc. About that same time, the echo cancelling mechanism (for the x100p) was rewritten to sense the audio reflection (or echo) during the first half-second or so of an initial pstn call. (That was a substantial improvement over previous cancellation methods without a doubt. If I recall recorrectly, that mechanism was reduced to sending an outbound short duration pulse or burst, and measuring the reflected energy. Sort of a snapshot at the start of an analog call. It's okay, but certainly not the equivalent of commercial analog cancellation products including mux's.) I've not had to revisit the x100p gain adjustment effort for several months, but seems to me that it was necessary to completely stop and start * each time an adjustment was made to the rxgain/txgain settings in zapata.conf (a simple reload wasn't adequate). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Intel 536ep as a FXO?
Hmmm...that doesn't work in my area either. I'm in the 509 area code, 448 exchange with Qwest and dialing 509-448-9996 gave me the no service announcement. Perhaps calling Qwest customer service and asking for the milliwat test number for my local calling area? Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis Sent: Monday, April 19, 2004 9:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? On Mon, 19 Apr 2004, Jeremy Hall wrote: This may not be the case in all areas, but in my area with Qwest as well, all exchanges have the test at xxx-9996. For example, my number is in the 208 area code, 459 exchange, so the full number would be 208-459-9996. It is not tied to any specific number, so I can use any exchange local to me such as 323-9996. It may or may not work in your area, so try not to do it at 3:00 AM until you have verified the number. I'm also in a Qwest area, but that number doesn't work here. All of the techs that I have asked gave it to me with no problems. They are shy about the automatic ANI number, however... dave -Original Message- From: Ed Rubright [mailto:[EMAIL PROTECTED] Sent: Monday, April 19, 2004 9:51 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? The next question for me is: How do I found out my telco milliwatt test number? I'm in Washington State using Qwest. The way I understand this, I'm to dialup the telco milliwatt test number and adjust the rxgain values using ztmonitor tool until the Max Audio Hit is in the middle of the bar graph for a normal conversation? Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, April 19, 2004 9:01 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? I for one would love this. I do not have any test equipment to determine the level I am sending at, but if I could at least figure out what levels to have my rxgain values set to, that would help. I remember seeing somewhere that you can use a program (part of the zt suite if I remember correctly) to view the audio levels on the FXO card like an on-screen vu meter. I can use that and dial up my telco milliwatt test number and adjust accordingly. I asked where that tool was on the IRC channel, but they seemed to not know either. I have searched as I know I saw it, but can't find it again. The tool you're looking for is /usr/src/zaptel/ztmonitor [EMAIL PROTECTED] zaptel]# ./ztmonitor Usage: ztmonitor channel num [-v] [-f FILE] [EMAIL PROTECTED] zaptel]# ./ztmonitor 1 -v Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ##* Keep in mind that tool is nothing more then an audio VU meter and was not intended to be an accurate means of measuring transmission levels. I think bkw (probably with Mark) wrote it back in the November/December timeframe as a simple tool for adjusting rxgain, etc. About that same time, the echo cancelling mechanism (for the x100p) was rewritten to sense the audio reflection (or echo) during the first half-second or so of an initial pstn call. (That was a substantial improvement over previous cancellation methods without a doubt. If I recall recorrectly, that mechanism was reduced to sending an outbound short duration pulse or burst, and measuring the reflected energy. Sort of a snapshot at the start of an analog call. It's okay, but certainly not the equivalent of commercial analog cancellation products including mux's.) I've not had to revisit the x100p gain adjustment effort for several months, but seems to me that it was necessary to completely stop and start * each time an adjustment was made to the rxgain/txgain settings in zapata.conf (a simple reload wasn't adequate). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dave Weis I believe there are more instances of the abridgment [EMAIL
RE: [Asterisk-Users] Intel 536ep as a FXO?
For the record, the milliwatt generator, ANI number, etc, is up to each telco engineering/operations group as to what number to assign to it. There are no industry standards at all. Since the xx98 and xx99 numbers use to be reserved for testing years ago, those numbers are still in frequent use. Also, some telco's use numbers like 311 for things like this, however the 411, 511, 611, 911 range has been filling up rather rapidly with other public things, so probably not to likely anymore. Easiest way to find them is to call Repair and ask. If that person can't tell you, ask for their supervisor. If that doesn't work, the next time you see a telephone truck, ask the driver; he's likely to be an employee that uses it more frequently then most others. Rich On Mon, 19 Apr 2004, Jeremy Hall wrote: This may not be the case in all areas, but in my area with Qwest as well, all exchanges have the test at xxx-9996. For example, my number is in the 208 area code, 459 exchange, so the full number would be 208-459-9996. It is not tied to any specific number, so I can use any exchange local to me such as 323-9996. It may or may not work in your area, so try not to do it at 3:00 AM until you have verified the number. I'm also in a Qwest area, but that number doesn't work here. All of the techs that I have asked gave it to me with no problems. They are shy about the automatic ANI number, however... dave -Original Message- From: Ed Rubright [mailto:[EMAIL PROTECTED] Sent: Monday, April 19, 2004 9:51 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? The next question for me is: How do I found out my telco milliwatt test number? I'm in Washington State using Qwest. The way I understand this, I'm to dialup the telco milliwatt test number and adjust the rxgain values using ztmonitor tool until the Max Audio Hit is in the middle of the bar graph for a normal conversation? Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, April 19, 2004 9:01 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? I for one would love this. I do not have any test equipment to determine the level I am sending at, but if I could at least figure out what levels to have my rxgain values set to, that would help. I remember seeing somewhere that you can use a program (part of the zt suite if I remember correctly) to view the audio levels on the FXO card like an on-screen vu meter. I can use that and dial up my telco milliwatt test number and adjust accordingly. I asked where that tool was on the IRC channel, but they seemed to not know either. I have searched as I know I saw it, but can't find it again. The tool you're looking for is /usr/src/zaptel/ztmonitor [EMAIL PROTECTED] zaptel]# ./ztmonitor Usage: ztmonitor channel num [-v] [-f FILE] [EMAIL PROTECTED] zaptel]# ./ztmonitor 1 -v Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ##* Keep in mind that tool is nothing more then an audio VU meter and was not intended to be an accurate means of measuring transmission levels. I think bkw (probably with Mark) wrote it back in the November/December timeframe as a simple tool for adjusting rxgain, etc. About that same time, the echo cancelling mechanism (for the x100p) was rewritten to sense the audio reflection (or echo) during the first half-second or so of an initial pstn call. (That was a substantial improvement over previous cancellation methods without a doubt. If I recall recorrectly, that mechanism was reduced to sending an outbound short duration pulse or burst, and measuring the reflected energy. Sort of a snapshot at the start of an analog call. It's okay, but certainly not the equivalent of commercial analog cancellation products including mux's.) I've not had to revisit the x100p gain adjustment effort for several months, but seems to me that it was necessary to completely stop and start * each time an adjustment was made to the rxgain/txgain settings in zapata.conf (a simple reload wasn't adequate). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] Intel 536ep as a FXO?
Also, while you have that phone guy cornered, you might try and get the ANI number - the one that reads back the number you're calling from. Quite useful if you're in the phone connection closet trying to locate your pair. Mine is 959-1122 (650 area code) Cheers! Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, April 19, 2004 10:49 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? For the record, the milliwatt generator, ANI number, etc, is up to each telco engineering/operations group as to what number to assign to it. There are no industry standards at all. Since the xx98 and xx99 numbers use to be reserved for testing years ago, those numbers are still in frequent use. Also, some telco's use numbers like 311 for things like this, however the 411, 511, 611, 911 range has been filling up rather rapidly with other public things, so probably not to likely anymore. Easiest way to find them is to call Repair and ask. If that person can't tell you, ask for their supervisor. If that doesn't work, the next time you see a telephone truck, ask the driver; he's likely to be an employee that uses it more frequently then most others. Rich On Mon, 19 Apr 2004, Jeremy Hall wrote: This may not be the case in all areas, but in my area with Qwest as well, all exchanges have the test at xxx-9996. For example, my number is in the 208 area code, 459 exchange, so the full number would be 208-459-9996. It is not tied to any specific number, so I can use any exchange local to me such as 323-9996. It may or may not work in your area, so try not to do it at 3:00 AM until you have verified the number. I'm also in a Qwest area, but that number doesn't work here. All of the techs that I have asked gave it to me with no problems. They are shy about the automatic ANI number, however... dave -Original Message- From: Ed Rubright [mailto:[EMAIL PROTECTED] Sent: Monday, April 19, 2004 9:51 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? The next question for me is: How do I found out my telco milliwatt test number? I'm in Washington State using Qwest. The way I understand this, I'm to dialup the telco milliwatt test number and adjust the rxgain values using ztmonitor tool until the Max Audio Hit is in the middle of the bar graph for a normal conversation? Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, April 19, 2004 9:01 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intel 536ep as a FXO? I for one would love this. I do not have any test equipment to determine the level I am sending at, but if I could at least figure out what levels to have my rxgain values set to, that would help. I remember seeing somewhere that you can use a program (part of the zt suite if I remember correctly) to view the audio levels on the FXO card like an on-screen vu meter. I can use that and dial up my telco milliwatt test number and adjust accordingly. I asked where that tool was on the IRC channel, but they seemed to not know either. I have searched as I know I saw it, but can't find it again. The tool you're looking for is /usr/src/zaptel/ztmonitor [EMAIL PROTECTED] zaptel]# ./ztmonitor Usage: ztmonitor channel num [-v] [-f FILE] [EMAIL PROTECTED] zaptel]# ./ztmonitor 1 -v Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ##* Keep in mind that tool is nothing more then an audio VU meter and was not intended to be an accurate means of measuring transmission levels. I think bkw (probably with Mark) wrote it back in the November/December timeframe as a simple tool for adjusting rxgain, etc. About that same time, the echo cancelling mechanism (for the x100p) was rewritten to sense the audio reflection (or echo) during the first half-second or so of an initial pstn call. (That was a substantial improvement over previous cancellation methods without a doubt. If I recall recorrectly, that mechanism was reduced to sending an outbound short duration pulse or burst, and measuring the reflected energy. Sort of a snapshot at the start of an analog call. It's okay, but certainly not the equivalent of commercial analog cancellation products including mux's.) I've not had to revisit the x100p gain adjustment effort for several months, but seems to me that it was necessary to
[Asterisk-Users] zaphfc
Hello list, I'm trying to use zaphfc, the module loads ok, and it identifies the hfc boards in the machine. The problem is: whenever i try to ztcfg -vv I get the following: 8x--- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) 8x-- when I try to start * it bails out with: == Parsing '/etc/asterisk/zapata.conf': Found Apr 19 17:27:34 WARNING[16384]: chan_zap.c:671 zt_open: Unable to specify channel 1: No such device or address Apr 19 17:27:34 ERROR[16384]: chan_zap.c:5338 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Apr 19 17:27:34 ERROR[16384]: chan_zap.c:7490 setup_zap: Unable to register channel '1' Apr 19 17:27:34 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 Apr 19 17:27:34 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! Junk at the beginning 49443303 Can anyone out there using zaphfc, help me on this? Thanks in advance, --- Paulo Loureiro. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about prepaid db
Hello, Somebody has an example with all data loaded in the base for prepaid? or an example of a base that this working?... Thanks... Julio
[Asterisk-Users] SIP call between 2 *
Hello, I'm new to the list and new to Asterisk. I'd like to know if any one has experience or configure files that can help me setup 2 * using SIP instead of IAX. I'm able to configure the * using IAX now and like to try SIP instead. Thanks! --- Tx Tim. __ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25¢ http://photos.yahoo.com/ph/print_splash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Does anyone now an Asterisk consultant in Atlanta? Bobby ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On Mon, 2004-04-19 at 14:22, Bobby Whitley wrote: Does anyone now an Asterisk consultant in Atlanta? Start Here, http://www.voip-info.org/wiki-Asterisk+consultants+USA -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] (no subject)
Does anyone now an Asterisk consultant in Atlanta? 1. Use the subject line - it's there for a reason. (no subject) won't draw too many people to read your message. 2. The wiki is your friend. See the URL below. There's no one listed for Atlanta, maybe that's why you're asking.. but see point 1 above ;-) http://www.voip-info.org/wiki-Asterisk+consultants+USA Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about prepaid db
Somebody made run prepaid?... - Original Message - From: Julio To: [EMAIL PROTECTED] Sent: Monday, April 19, 2004 2:38 PM Subject: [Asterisk-Users] Question about prepaid db Hello, Somebody has an example with all data loaded in the base for prepaid? or an example of a base that this working?... Thanks... Julio
[Asterisk-Users] One, två, tre, quatre, cinq ... International numbers in say.c
http://bugs.digium.com/bug_view_page.php?bug_id=0001429 * Support for other language syntaxes in saynumber Accidentally I opened this can of worms to see if we can add support for other language syntaxes for saying numbers. Seems like Swedish, english and norwegian follow the same syntax. I've integrated existing patches for french, danish and soon portuguese syntax. The steps we're taking are: * First a quick-fix only for saying numbers * Adding documentation and sample sound files Many patches require additional sound files compared with the english set. * For a coming release we need a more general architecture that includes more phrases, time and date. This will be done with loadable modules for various languages. I need the original contributors of danish, french and portuguese to fax a disclaimer to Digium. See http://bugs.digium.com Also, I need users in these language territories to test the patch and add feedback to the bugtracker. I can try to put all this together into one unified patch, but not test everything for every language. If you have a patch for another syntax, please add it quickly to the bugtracker and fax in the disclaimer, so we can use it. If you have sound files for a language with decent quality that you can share to the community, please do so by adding them to the bug tracker. * If we all work on this together quickly, we may have a working say.c in the CVS soon. But to even ask a committer for support, I need test results up there on the bug tracker. * Thank you for your support! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playback problems with T100P
Hi, I'm seeing a problem when calling into my Asterisk server from a T100P with PRI signalling. I'm hoping someone has seen this as well, however, it's a little hard to explain. When I'm navigating through Asterisk on an inbound call from the PSTN, I'm getting a random 'noise' during the playback of sound files. The best way I can describe this noise, is imagining a windows machine crashing while it's playing back PCM. The sound card holds onto the last few milliseconds of data in it's buffer and plays it in a loop, creating a (rather painful to the ears) tone. Another analogy is the age old pulling out an Atari cartridge sound. The noise lasts for 2 or 3 seconds ceases. The sound playback continues afterwards and was never interrupted, or stopped, for the duration of this noise. As stated above, this happens at random. I've tested the line myself with a berd and I have no errors or slips. I've ruled that part out. I'm not sure if the problem is related to the T100P itself, or if it's caused by the PC hardware, OS, or Asterisk. I'd like to rule as much out as I can before calling Digium about the hardware. Any help is appreciated. Thank you. - Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One, två, tre, quatre, cinq ... International numbers in say.c
hi Olle. I have a patch for italian. should it be for plain say.c or for your modified say.c ? Also I have some the .it audio files, I'll ask if I can distribute them (perhaps with some credit to the company I work for, that payed them...) Matteo Il lun, 2004-04-19 alle 21:53, Olle E. Johansson ha scritto: http://bugs.digium.com/bug_view_page.php?bug_id=0001429 * Support for other language syntaxes in saynumber Accidentally I opened this can of worms to see if we can add support for other language syntaxes for saying numbers. Seems like Swedish, english and norwegian follow the same syntax. I've integrated existing patches for french, danish and soon portuguese syntax. The steps we're taking are: * First a quick-fix only for saying numbers * Adding documentation and sample sound files Many patches require additional sound files compared with the english set. * For a coming release we need a more general architecture that includes more phrases, time and date. This will be done with loadable modules for various languages. I need the original contributors of danish, french and portuguese to fax a disclaimer to Digium. See http://bugs.digium.com Also, I need users in these language territories to test the patch and add feedback to the bugtracker. I can try to put all this together into one unified patch, but not test everything for every language. If you have a patch for another syntax, please add it quickly to the bugtracker and fax in the disclaimer, so we can use it. If you have sound files for a language with decent quality that you can share to the community, please do so by adding them to the bug tracker. * If we all work on this together quickly, we may have a working say.c in the CVS soon. But to even ask a committer for support, I need test results up there on the bug tracker. * Thank you for your support! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One, två, tre, quatre, cinq ... International numbers in say.c
Brancaleoni Matteo wrote: hi Olle. I have a patch for italian. Great. should it be for plain say.c or for your modified say.c ? If you have one that builds on my patch, that'll make life easier for me. THank you! Also I have some the .it audio files, I'll ask if I can distribute them (perhaps with some credit to the company I work for, that payed them...) Of course. Anyone else? This seems so focused on Western Europe? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback problems with T100P
On Mon, 2004-04-19 at 14:59, Eric Einhorn wrote: Hi, I'm seeing a problem when calling into my Asterisk server from a T100P with PRI signalling. I'm hoping someone has seen this as well, however, it's a little hard to explain. When I'm navigating through Asterisk on an inbound call from the PSTN, I'm getting a random 'noise' during the playback of sound files. The best way I can describe this noise, is imagining a windows machine crashing while it's playing back PCM. The sound card holds onto the last few milliseconds of data in it's buffer and plays it in a loop, creating a (rather painful to the ears) tone. Another analogy is the age old pulling out an Atari cartridge sound. The noise lasts for 2 or 3 seconds ceases. The sound playback continues afterwards and was never interrupted, or stopped, for the duration of this noise. As stated above, this happens at random. I've tested the line myself with a berd and I have no errors or slips. I've ruled that part out. I'm not sure if the problem is related to the T100P itself, or if it's caused by the PC hardware, OS, or Asterisk. I'd like to rule as much out as I can before calling Digium about the hardware. Any help is appreciated. You've done a great job describing your problem with the exception of documenting all the hardware in the system and software versions. As a way of eliminating some of the questionable parts, you must enumerate that part of your setup. Also, where is your T100P pointing to, telco, pbx, or some other hardware? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc
Hello, Can you post zapata.conf and zaptel.conf ? It's seems a config file problem. At 19:32 19/04/2004, you wrote: Hello list, I'm trying to use zaphfc, the module loads ok, and it identifies the hfc boards in the machine. The problem is: whenever i try to ztcfg -vv I get the following: 8x--- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) 8x-- when I try to start * it bails out with: == Parsing '/etc/asterisk/zapata.conf': Found Apr 19 17:27:34 WARNING[16384]: chan_zap.c:671 zt_open: Unable to specify channel 1: No such device or address Apr 19 17:27:34 ERROR[16384]: chan_zap.c:5338 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Apr 19 17:27:34 ERROR[16384]: chan_zap.c:7490 setup_zap: Unable to register channel '1' Apr 19 17:27:34 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 Apr 19 17:27:34 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! Junk at the beginning 49443303 Can anyone out there using zaphfc, help me on this? Thanks in advance, --- Paulo Loureiro. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] capi_request: didn't find capi device with outgoing msn =
Hi,
I can't make outgoing calls with CAPI (passive ISDN Fritz card). See
Asterisk error below.
Incoming calls and SIP to SIP calls do work. It looks like a msn
mismatch in extensions.conf
and capi.conf, but I can't find it.
Can anyone help me find the problem?
Thanks,
Rob
*CLI
-- Executing Dial(SIP/8112-1be9, CAPI/35666:BYEXTENSION) in
new stack
Apr 19 13:00:11 NOTICE[671760]: chan_capi.c:1147 capi_request: didn't
find capi device with outgoing msn = 35666. you should check your
config!
Apr 19 13:00:11 NOTICE[671760]: app_dial.c:554 dial_exec: Unable to
create channel of type 'CAPI'
== Everyone is busy at this time
-- Executing Congestion(SIP/8112-1be9, ) in new stack
== Spawn extension (home, 035999, 2) exited non-zero on
'SIP/8112-1be9'
*CLI
*CLIcapi info
Contr1: 2 B channels total, 2 B channels free.
*CLI
capi.conf:
msn=35666
modem.conf:
msn=35666
extensions.conf:
exten = _0.,1,Dial,${TRUNK}/35666:BYEXTENSION
exten = _0.,2,Congestion
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[Asterisk-Users] IAX config documentation
Is there any documentation on configuring IAX between * machines? I've noticed references to many topics in the config files, including: - dialplans - trunking - authentication - transfers But before I go and try to grok 8000 lines of source (in one file, no less) I was hoping that somewhere there exists even something like a man page that describes the configuration options. There's the beginnings of a whitepaper on wiki, but it's self-contradictory in some places, largely incomplete, and just kind of ends abruptly. Yet, it mentions that growing contingent of IAX devices. Huh. Thanks -brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capi_request: didn't find capi device with outgoing msn =
Rob wrote: I can't make outgoing calls with CAPI (passive ISDN Fritz card). See Asterisk error below. Incoming calls and SIP to SIP calls do work. It looks like a msn mismatch in extensions.conf and capi.conf, but I can't find it. I had the same problem. A reboot of the system solved it. hth rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX config documentation
On Mon, 2004-04-19 at 16:08, Brian Cuthie wrote: Is there any documentation on configuring IAX between * machines? I've noticed references to many topics in the config files, including: - dialplans This is extensions.conf work. Some of it can be shared via the switch command. - trunking Trunking is easy, think of it kind of like a channelized t1. It combines many calls into one packet with call data so as to reduce the overhead of each individual call having it's own resources. Specifically it cuts down on the overhead in IP, and allows you to reclaim some of the bandwidth for more calls. - authentication You do want to know who is trying to call you don't you? - transfers Allows you to get out of the middle of a call. My office loves these as our trunk lines are remote, and when we forward a call out to another trunk line, our local asterisk machine transfers the call back to the machine with trunk lines and removes the VoIP part of the loop. But before I go and try to grok 8000 lines of source (in one file, no less) I was hoping that somewhere there exists even something like a man page that describes the configuration options. There is plenty of banter on the list and info scattered about that google will find for you than reading the source. Of course, you are free to bludgen yourself with the code if you so wish. There's the beginnings of a whitepaper on wiki, but it's self-contradictory in some places, largely incomplete, and just kind of ends abruptly. Yet, it mentions that growing contingent of IAX devices. Huh. Jump in and help finish it when you have read some and start to understand the missing parts. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk
Dear List, I've compiled asterisk (both 0.9.0 and the CVS-04/19/04 source trees). I'm using the oh323 channel driver version 0.5.10, OpenH323 v1.12.2, PWlib v1.5.2 When run on a RedHat 9 system, I am constantly getting seg faults. This happens even when I tried removing the oh323 channel driver, so it appears to be something with asterisk. I get crashes either when attempting to start asterisk or when asterisk receives an incoming h323 call. When run on a RedHat 7.3 system (exact same source code) both asterisk and the oh323 channel driver appear to be stable. Does anyone have any advice? I assume this has something to do with incompatible libraries, but have no idea where to start. TIA Chris -- Chris Wik Systems Admin ANU Internet Services http://www.anu.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Mac OS X 10.3
Hi, Has anybody successfully compiled Asterisk on Mac OS X 10.3 (Panther)? If so, I would be grateful for instructions on how to achieve a successful installation. I am running on an Apple G4 dual processor. Thanks, Steve Macartney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Load module chan_zap.so failed
Hi I' ve just installed TE410P and asterisk-0.7.2 from tar.gz on fedora core 1. When i start asterisk it shows me this: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 19 16:52:32 WARNING[-1085304704]: loader.c:358 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance Jorge Verastegui G RedCetus S.R.L --NOTA DE REDCETUS S.R.L. : La informacin contenida en este E-mail y sus anexos, slo puede ser utilizada por el individuo o la compaia a la cual est dirigido. Si no es el receptor autorizado, cualquier retencin, difusin, distribucin o copia de este mensaje es prohibida y sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo y borrar el mismo
[Asterisk-Users] queue out
Please: There is some form so that a user in the queue leaves her (with a digit) and the system execute another command (for example goto a voice mailbox). My version: Asterisk CVS-04/16/04 Thanks in advance -- Jose Mª Guisasola Consultor Técnico CMSI 2002 S.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp/rxfax terminates asterisk
Initial handshake sounds fine, but asterisks dies before receive of the fax. Here is the log : Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 30 36 37 37 36 31 36 35 20 35 34 2b 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: +45 56167760 DCS: 83 00 46 20 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 10ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1700.00 (64) Training error 29.095569 Training succeeded (constellation mismatch 25.504344) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 1 Start rx page asterisk in realloc(): warning: junk pointer, too high to make sense Oh dear! CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Coarse carrier frequency 1700.04 (64) Training error 26.487284 Training succeeded (constellation mismatch 27.123313) Fast carrier trained Segmentation fault (core dumped) Anyone ? Thanks, Martin Min mail er beskyttet af SPAMfighter 3174 spam mails er blokeret indtil videre.Hent gratis SPAMfighter i dag!
Re: [Asterisk-Users] Load module chan_zap.so failed
you must ztcfg -vv then modprobe your zaptel harware... jorge verastegui wrote: Hi I' ve just installed TE410P and asterisk-0.7.2 from tar.gz on fedora core 1. When i start asterisk it shows me this: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 19 16:52:32 WARNING[-1085304704]: loader.c:358 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance Jorge Verastegui G RedCetus S.R.L -- *NOTA DE REDCETUS S.R.L.* : La información contenida en este E-mail y sus anexos, sólo puede ser utilizada por el individuo o la compañia a la cual está dirigido. Si no es el receptor autorizado, cualquier retención, difusión, distribución o copia de este mensaje es prohibida y sancionada por la ley. Si por error recibe este mensaje, favor reenviarlo y borrar el mismo -- Todd Lieberman http://tlsolutions.net mailto:[EMAIL PROTECTED] p. 215.495.0030 f. 215.495.0031 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ixj module
Hello list: If I am not mistaken the module 'ixj' is for the cards 'Quicknet LineJack' is possible not to load it when starting asterisk ?. How call asterisk this module ?. -- Jose Mª Guisasola Consultor Técnico CMSI 2002 S.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap Outgoing
Hello All, i m having busy signal when i dial any number, while incoming on zap is working fine and its transfering to my soft phone. some time back outgoing was working ok but now i dont know what i messed up. any idea ? it gives busy signal after Zap/25-1 answered SIP/300 -Neo = Spawn extension (voicepulse-incoming, s, 1) exited non-zero on 'Zap/25-1' -- Hungup 'Zap/25-1' -- Executing Dial(SIP/3000-2e72, Zap/25/18005558355) in new stack -- Called 25/18005558355 -- Zap/25-1 answered SIP/3000-2e72 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk prepaid debug
My Asterisk prepaid debug is: - Hungup 'Zap/2-1'Urgent handler -- Starting simple switch on 'Zap/2-1'Urgent handler -- Playing 'prepaid-enter-card-num' (language 'en')Urgent handler -- Playing 'prepaid-you-have' (language 'en')Urgent handler -- Playing 'digits/4' (language 'en')Urgent handler -- Playing 'digits/hundred' (language 'en')Urgent handler -- Playing 'prepaid-dollars' (language 'en')Urgent handler -- Playing 'prepaid-enter-dest' (language 'en')Urgent handler -- Playing 'prepaid-dest-blocked' (language 'en')Urgent handler -- Playing 'prepaid-dest-unreachable' (language 'en') Why 'prepaid-dest-unreachable' ?? Thks. Regards - Original Message - From: Martin Christian Koch To: [EMAIL PROTECTED] Sent: Monday, April 19, 2004 4:05 PM Subject: [Asterisk-Users] spandsp/rxfax terminates asterisk Initial handshake sounds fine, but asterisks dies before receive of the fax. Here is the log : Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 30 36 37 37 36 31 36 35 20 35 34 2b 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: "+45 56167760" DCS: 83 00 46 20 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 10ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1700.00 (64) Training error 29.095569 Training succeeded (constellation mismatch 25.504344) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 1 Start rx page asterisk in realloc(): warning: junk pointer, too high to make sense Oh dear! CFR: 84 HDLC underflow in
[Asterisk-Users] Random Disconnects
I am getting random disconnects about 5-10 times a day. The logs show nothing except that the call was hung up. The calls are from X100P-*-digium T1 card-carrier access channel bank II-analogue phone. It is happening to all users. Is it possible that this is coming from busydetect=yes? Does busydetect detect cadences etc for the hangup frequencies? I have busycount=3... Any ideas? Any more information I could provide? Kind regards, Matt Riddell
[Asterisk-Users] -- MARK --
Every half hour I get -- MARK -- in the syslog. Is this normal behavior? Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] -- MARK --
Every half hour I get -- MARK -- in the syslog. Is this normal behavior? Yup - I get it too, although I seem to remember it was more of a Slackware thing than a RedHat thing.. I think it's configurable too, so you can turn it off if it's pissing you off. Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] -- MARK --
On Mon, 19 Apr 2004, Michael Welter wrote: Every half hour I get -- MARK -- in the syslog. Is this normal behavior? This has nothing to be with asterisk, but with your linux installation. Yes, it is a normal behavior and it is harmless... It is just a half hour stamp to your syslog... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] -- MARK --
From syslogd man page: -m interval The syslogd logs a mark timestamp regularly. The default inter-val between two -- MARK -- lines is 20 minutes. This can be changed with this option. Setting the interval to zero turns it off entirely. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Monday, April 19, 2004 7:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] -- MARK -- On Mon, 19 Apr 2004, Michael Welter wrote: Every half hour I get -- MARK -- in the syslog. Is this normal behavior? This has nothing to be with asterisk, but with your linux installation. Yes, it is a normal behavior and it is harmless... It is just a half hour stamp to your syslog... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects
Hi Matt, Increase your busycount to 6 or 7. I had that problem also with an X100P, and it went away increasing the busycount parameter. On Mon, 2004-04-19 at 20:28, Matt Riddell wrote: I am getting random disconnects about 5-10 times a day. The logs show nothing except that the call was hung up. The calls are from X100P-*-digium T1 card-carrier access channel bank II-analogue phone. It is happening to all users. Is it possible that this is coming from busydetect=yes? Does busydetect detect cadences etc for the hangup frequencies? I have busycount=3... Any ideas? Any more information I could provide? Kind regards, Matt Riddell -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] -- MARK --
On Mon, 19 Apr 2004, Michael Welter wrote: Every half hour I get -- MARK -- in the syslog. Is this normal behavior? This has nothing to be with asterisk, but with your linux installation. Yes, it is a normal behavior and it is harmless... It is just a half hour stamp to your syslog... I think it was because of MARK Spencer... burn him! :-) Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users] One, två, tre, quatre, cinq ... International numbers in say.c
I have the sounds for French, but can record more if necessary. They are available at www.sineapps.com Is a disclaimer required on these? Kind regards, Matt Riddell - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: Users Asterisk [EMAIL PROTECTED] Sent: Tuesday, April 20, 2004 7:53 AM Subject: [Asterisk-Users] One, två, tre, quatre, cinq ... International numbers in say.c | http://bugs.digium.com/bug_view_page.php?bug_id=0001429 | | * Support for other language syntaxes in saynumber | | Accidentally I opened this can of worms to see if we can add support | for other language syntaxes for saying numbers. Seems like Swedish, | english and norwegian follow the same syntax. I've integrated | existing patches for french, danish and soon portuguese syntax. | | The steps we're taking are: | | * First a quick-fix only for saying numbers | * Adding documentation and sample sound files |Many patches require additional sound files compared with the |english set. | * For a coming release we need a more general architecture that |includes more phrases, time and date. This will be done with |loadable modules for various languages. | | I need the original contributors of danish, french and portuguese | to fax a disclaimer to Digium. See http://bugs.digium.com | | Also, I need users in these language territories to test the | patch and add feedback to the bugtracker. I can try to put all this | together into one unified patch, but not test everything for every | language. | | | If you have a patch for another syntax, please add it quickly to | the bugtracker and fax in the disclaimer, so we can use it. | | If you have sound files for a language with decent quality that | you can share to the community, please do so by adding them to | the bug tracker. | | * If we all work on this together quickly, we may have a | working say.c in the CVS soon. But to even ask a committer for | support, I need test results up there on the bug tracker. * | | Thank you for your support! | | /Olle | | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects
- Original Message - | Hi Matt, | | Increase your busycount to 6 or 7. I had that problem also with an | X100P, and it went away increasing the busycount parameter. Now that I check it, I don't have a busycount...does this really need to be set in dsp.c? If so how would I compile it and install it with the machine running? Cheers, Matt | On Mon, 2004-04-19 at 20:28, Matt Riddell wrote: | I am getting random disconnects about 5-10 times a day. The logs show | nothing except that the call was hung up. The calls are from | X100P-*-digium T1 card-carrier access channel bank II-analogue | phone. It is happening to all users. Is it possible that this is | coming from busydetect=yes? | | Does busydetect detect cadences etc for the hangup frequencies? I | have busycount=3... | | Any ideas? Any more information I could provide? | | Kind regards, | | | Matt Riddell | -- | Nicolas Gudino [EMAIL PROTECTED] | House Internet S.R.L. | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects
| - Original Message - | | Hi Matt, | | | | Increase your busycount to 6 or 7. I had that problem also with an | | X100P, and it went away increasing the busycount parameter. | | Now that I check it, I don't have a busycount...does this really need to be | set in dsp.c? | | If so how would I compile it and install it with the machine running? | According to http://www.automated.it/guidetoasterisk.htm it goes in the zapata.conf file after busydetect=yes. Soz for mail list spam... Kind regards, Matt Riddell | | On Mon, 2004-04-19 at 20:28, Matt Riddell wrote: | | I am getting random disconnects about 5-10 times a day. The logs show | | nothing except that the call was hung up. The calls are from | | X100P-*-digium T1 card-carrier access channel bank II-analogue | | phone. It is happening to all users. Is it possible that this is | | coming from busydetect=yes? | | | | Does busydetect detect cadences etc for the hangup frequencies? I | | have busycount=3... | | | | Any ideas? Any more information I could provide? | | | | Kind regards, | | | | | | Matt Riddell | | -- | | Nicolas Gudino [EMAIL PROTECTED] | | House Internet S.R.L. | | | | ___ | | Asterisk-Users mailing list | | [EMAIL PROTECTED] | | http://lists.digium.com/mailman/listinfo/asterisk-users | | To UNSUBSCRIBE or update options visit: | |http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP SIP SoftPhone Recommendations
Can you give me the configuration archives, I have a problems witch clients on *. Thanx. JRR From: Edmund [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoIP SIP SoftPhone Recommendations Date: Mon, 19 Apr 2004 10:50:17 +0800 I'm using Linphone. It works pefectly with *. Edmund JORA ROME wrote: What SoftPhone working very well with *? S.O. is Debian Linux Thanks for your comments. JRR _ MSN Amor: busca tu ½ naranja http://latam.msn.com/amor/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN Amor: busca tu ½ naranja http://latam.msn.com/amor/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Coredump while txfax - case2
Title: Coredump while txfax - case2 Hi Steve and all, This is the 2nd attachment. In many cases txfax works on our asterisk with RH9, spandsp.0.0.1k and libtiff.so.3.5. The attached tif file and another one (case 2, attached to the next message) crash * consistently in libtiff. Both files had been received by rxfax. ... Changed from phase 3 to 6 Changed from phase 6 to 4 Start tx page Page 3 of /usr/tmp/susp2.tif 128 rows/1866 bytes to send MPS: 4f HDLC underflow in state 13 Changed from phase 4 to 3 Slow carrier up MCF: 8c MCF with final frame tag In state 13 Changed from phase 3 to 6 Changed from phase 6 to 4 Start tx page #0 0x4084398c in TIFFWriteBufferSetup () from /usr/local/lib/libtiff.so.3 (gdb) bt #0 0x4084398c in TIFFWriteBufferSetup () from /usr/local/lib/libtiff.so.3 #1 0x40843b08 in TIFFFlushData1 () from /usr/local/lib/libtiff.so.3 #2 0x4082fab4 in _TIFFFax3fillruns () from /usr/local/lib/libtiff.so.3 #3 0x408303e8 in _TIFFFax3fillruns () from /usr/local/lib/libtiff.so.3 #4 0x40830589 in _TIFFFax3fillruns () from /usr/local/lib/libtiff.so.3 #5 0x4080e666 in t4_tx_start_page () from /usr/local/lib/libspandsp.so.0 #6 0x4080eb06 in fast_getbit () from /usr/local/lib/libspandsp.so.0 #7 0x40819617 in getbaud () from /usr/local/lib/libspandsp.so.0 #8 0x408198ad in v29_tx () from /usr/local/lib/libspandsp.so.0 #9 0x4081289f in fax_tx_process () from /usr/local/lib/libspandsp.so.0 #10 0x4086539a in txfax_exec (chan=0x81b2ed8, data="" at app_txfax.c:216 #11 0x0806377a in pbx_exec (c=0x81b2ed8, app=0x81aa9f0, data="" newstack=1) at pbx.c:396 #12 0x0806ac81 in pbx_extension_helper (c=0x81b2ed8, context=0x81b3030 webley_txfax, exten=0xbc7ff65c /usr/tmp/susp2.tif|caller, priority=2, callerid=0x0, action="" at pbx.c:1157 #13 0x0806568c in ast_pbx_run (c=0x81b2ed8) at pbx.c:1641 #14 0x080681b0 in ast_pbx_outgoing_exten (type=0x81b1cc8 Zap, format=64, data="" timeout=12, context=0x81b21c8 webley_txfax, exten=0x81b20c8 txfax_ext, priority=1, reason=0xbc7ff65c, sync=135999192, callerid=0x81b2ed8 Zap/1-1, variable=0x81b23cc TXFAX_NAME, account=0x81b2dcc ) at pbx.c:3838 #15 0x406de2ab in attempt_thread (data="" at pbx_spool.c:196 #16 0x4003fae0 in pthread_start_thread () from /lib/libpthread.so.0 (gdb) Would be great to fix this one... Thank you. Alex Zarubin Webley Systems susp2.tif.gz Description: Binary data
RE: [Asterisk-Users] Accommodating multiple FWD users
Thanks to Philipp, Eric and Vic for their responses. In the end, I decided
to use Philipp's approach and it works like a charm!
Malcolm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: Monday, April 19, 2004 10:33 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Accommodating multiple FWD users
Hi!
Can anyone suggest a way in which all users could dial the prefix 8 and
*
would automatically associate the correct FWD account for the outbound
call?
Try using GoToIf [show application gotoif] in combination with
${CALLERIDNUM} [asterisk/doc/README.variables]
I prefer a slightly cleaner method:
1. create a type=peer entry for each outgoing FWD account in sip.conf.
For example you have [fwd-out_joe], [fwd-out_bob], [fwd-out_mary] where
you specify the individual username, fromuser and password
2. in sip.conf put each of your local phone users into their own context
like context=from-joe or context=from-mary
3. in extension.conf you do smth like include = default for each of
those person contexts like [from-joe], and arrange a FWD dialout like
_8X. = Dial(SIP/{EXTEN:[EMAIL PROTECTED])
If, however, you want to avoid individual contexts in extensions.conf you
could instead us DBput() and DBget() lookup the correct fwd-out_xxx
string based upon your local users's callerid.
Cheers, Philipp
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Re: [Asterisk-Users] IAX config documentation
I know that this stuff is. What I'm looking for is an overview of how these features work in the context of IAX. For instance, trunking is a concept I think we all get. But how do you use IAX to establish trunking between two switches? What's the effect of turning the transfer option on? How are dialplans shared between switches that are connected via IAX? What kinds of authentication are supported? How are keys managed? -brian Steven Critchfield wrote: On Mon, 2004-04-19 at 16:08, Brian Cuthie wrote: Is there any documentation on configuring IAX between * machines? I've noticed references to many topics in the config files, including: - dialplans This is extensions.conf work. Some of it can be shared via the switch command. - trunking Trunking is easy, think of it kind of like a channelized t1. It combines many calls into one packet with call data so as to reduce the overhead of each individual call having it's own resources. Specifically it cuts down on the overhead in IP, and allows you to reclaim some of the bandwidth for more calls. - authentication You do want to know who is trying to call you don't you? - transfers Allows you to get out of the middle of a call. My office loves these as our trunk lines are remote, and when we forward a call out to another trunk line, our local asterisk machine transfers the call back to the machine with trunk lines and removes the VoIP part of the loop. But before I go and try to grok 8000 lines of source (in one file, no less) I was hoping that somewhere there exists even something like a man page that describes the configuration options. There is plenty of banter on the list and info scattered about that google will find for you than reading the source. Of course, you are free to bludgen yourself with the code if you so wish. There's the beginnings of a whitepaper on wiki, but it's self-contradictory in some places, largely incomplete, and just kind of ends abruptly. Yet, it mentions that growing contingent of IAX devices. Huh. Jump in and help finish it when you have read some and start to understand the missing parts. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need Help with Dial Plan
Let me lay it out for you
Call comes in over a T1 - Signal is em_w. The extension is seen as
*callerid*last 4 digits of number being called*. Which is fine in
it self.
I have my extension.conf file set up as follows...
[did]
; Receive call as *calling*called
exten = _.,1,Answer
exten = _.,2,Cut(CALLING=EXTEN,*,2)
exten = _.,3,SetCIDNum(${CALLING})
exten = _.,4,Cut(CALLED=EXTEN,*,3)
exten = _.,5,Goto(main,${CALLED},1)
include = main
[main]
exten = 0031,1,Answer
exten = 0031,2,Goto(TNE-SG,s,1)
Include = did
include = TNE-SG
[TNE-SG]
exten = s,1,Answer
;exten = s,2,agi,tne.agi
exten = s,2,Background(tne-main-thanks)
exten = s,3,Background(tne-main-menu)
exten = 1,1,Goto(default-tne,9100,1)
exten = 2,1,Goto(default-tne,4100,1)
exten = 3,1,Goto(default-tne,4200,1)
exten = 4,1,Goto(default-tne,4300,1)
exten = 5,1,Goto(default-tne,4400,1)
exten = 6,1,Goto(tne-main-menu,s,3)
exten = 7,1,Hangup
include = default-tne
include = main
[default-tne]
include = TNE-SG
; Geoff Clark
exten = 4001,1,Macro(stdexten,4001,SIP/gclark)
;exten = 4001,1,Dial(IAX/home:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = 4004,1,Macro(stdexten,4004,SIP/home)
; Kyle Elworthy
exten = 4002,1,Macro(stdexten,4002,SIP/kelworth)
exten = 4003,1,Macro(stdexten,4003,SIP/khome)
; Tech Support Agents
exten = *6,1,AgentCallbackLogin(4001,[EMAIL PROTECTED])
exten = *7,1,AgentCallbackLogin(4002,[EMAIL PROTECTED])
exten = 401,1,Dial(Zap/g1/7046223905)
exten = 402,1,Dial(Zap/g1/7049071514)
exten = 411,1,Answer
exten = 411,2,Wait,2
exten = 411,3,Background(auth-thankyou)
exten = 411,4,Queue(tech-supp)
Where the problem comes in is - I can dial in fine in this scenerio -
but when I go to make an outbound call, it calls the did context and
cut's the call up.
My problem appears to be I need it one way but not the other.. I hope
this makes since...
Thanks,
-gcc
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Re: [Asterisk-Users] Asterisk prepaid debug
What kind of prepaid agi did you use ?Could you send me the page ? How install or where find ? Thanks alot. On Mon, 19 Apr 2004 20:07:14 -0600, Julio wrote: My Asterisk prepaid debug is: - Hungup 'Zap/2-1' Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler -- Playing 'prepaid-enter-card-num' (language 'en') Urgent handler -- Playing 'prepaid-you-have' (language 'en') Urgent handler -- Playing 'digits/4' (language 'en') Urgent handler -- Playing 'digits/hundred' (language 'en') Urgent handler -- Playing 'prepaid-dollars' (language 'en') Urgent handler -- Playing 'prepaid-enter-dest' (language 'en') Urgent handler -- Playing 'prepaid-dest-blocked' (language 'en') Urgent handler -- Playing 'prepaid-dest-unreachable' (language 'en') Why 'prepaid-dest-unreachable' ?? Thks. Regards - Original Message - From: Martin Christian Koch To: [EMAIL PROTECTED] Sent: Monday, April 19, 2004 4:05 PM Subject: [Asterisk-Users] spandsp/rxfax terminates asterisk Initial handshake sounds fine, but asterisks dies before receive of the fax. Here is the log : Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 30 36 37 37 36 31 36 35 20 35 34 2b 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: "+45 56167760" DCS: 83 00 46 20 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 10ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1700.00 (64) Training error 29.095569 Training succeeded (constellation mismatch 25.504344) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 1 Start rx page asterisk in realloc(): warning: junk pointer, too high to make sense Oh dear! CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Coarse carrier frequency 1700.04 (64) Training error 26.487284 Training succeeded (constellation mismatch 27.123313) Fast carrier trained Segmentation fault (core dumped) Anyone ? Thanks, Martin Min mail er beskyttet af SPAMfighter 3174 spam mails er blokeret indtil videre. Hent gratis SPAMfighter i dag!
[Asterisk-Users] Connecting PBX to Asterisk
Im trying to inter-connect my current PBX system and Asterisk. Asterisk has some users from different networks (internet).. I used cisco router using 4 fxs to pbx and SIP to asterisk. Is there any way i can allow the ip address of cisco to connect to my asterisk using SIP? IP Address of cisco is 192.168.0.254 here's a part of my sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = default srvlookup = yes pedantic = yes tos=lowdelay maxexpirey=360 defaultexpirey=120 disallow=all allow=ulaw allow=alaw [2101] type=friend context=sip-users secret= host=dynamic username=2101 qualify=yes nat=yes canreinvite=no and my extensions.conf [sip-users] exten =_21XX,1,Dial(SIP/[EMAIL PROTECTED]) [default] exten s,1,Hangup Regards, Antonio Rabena ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura line 1 outgoing voice problem?
I did some packet sniffs, and below are two sets of packets, the first is the second phone line that works fine with an incoming call and outgoing sound This seems to be the key packet that sets up the codes and sessions ( I really don't know any of this sip stuff well, but hopefully somebody on the list knows it): The main thing to point out is the initial Media Description section. In the Working line2, it's: Media Description, name and address (m): audio 16446 RTP/AVP 0 101 ... Media Format: 101 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute Fieldname: fmtp Media Attribute Value: 101 0-15 So I believe this is setting up the audio transmit stuff. On the line1, this data is NOT being sent. I don't know what this stuff means, still looking into it..but maybe someone here does know it? Am I possibly even on the right track?? The other thought I have, since this is data being sent FROM the Sipura TO asterisk, the problem is once again seeming to point directly at Sipura, and it's basically not sending the audio info.. Does any of this even make any sense?? Hope this either helps others to possibly find a fix, or if anyone _does_ have a fix, please let me know! Packet for line2, working outgoing audio Frame 7 (733 bytes on wire, 733 bytes captured) Ethernet II, Src: 00:0e:08:aa:b7:b1, Dst: 00:07:95:55:7b:ce Internet Protocol, Src Addr: 192.168.1.21 (192.168.1.21), Dst Addr: 192.168.1.20 (192.168.1.20) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) Session Initiation Protocol Status line: SIP/2.0 200 OK Status-Code: 200 Message Header To: sip:[EMAIL PROTECTED]:5061;tag=6b4e39bb53bc50bc SIP to address: sip:[EMAIL PROTECTED]:5061 SIP tag: 6b4e39bb53bc50bc From: asterisk sip:[EMAIL PROTECTED];tag=as55a02558 SIP from address: asterisk sip:[EMAIL PROTECTED] SIP tag: as55a02558 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK0d8c5d0f Contact: SPA 2202 sip:[EMAIL PROTECTED]:5061 Server: Sipura/SPA2000-2.0.2 Content-Length: 210 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 25669620 25669620 IN IP4 192.168.1.21 Owner Username: - Session ID: 25669620 Session Version: 25669620 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 192.168.1.21 Session Name (s): - Connection Information (c): IN IP4 192.168.1.21 Connection Network Type: IN Connection Address Type: IP4 Connection Address: 192.168.1.21 Time Description, active time (t): 0 0 Session Start Time: 0 Session Start Time: 0 Media Description, name and address (m): audio 16446 RTP/AVP 0 101 Media Type: audio Media Port: 16446 Media Proto: RTP/AVP Media Format: 0 Media Format: 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute Fieldname: fmtp Media Attribute Value: 101 0-15 Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20 Media Attribute (a): sendrecv Below is an incoming phone call to line1, with the outgoing voice NOT working: Frame 7 (672 bytes on wire, 672 bytes captured) Ethernet II, Src: 00:0e:08:aa:b7:b1, Dst: 00:07:95:55:7b:ce Internet Protocol, Src Addr: 192.168.1.21 (192.168.1.21), Dst Addr: 192.168.1.20 (192.168.1.20) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status line: SIP/2.0 200 OK Status-Code: 200 Message Header To: sip:[EMAIL PROTECTED];tag=f03d01bbf25c28bb SIP to address: sip:[EMAIL PROTECTED] SIP tag: f03d01bbf25c28bb From: asterisk sip:[EMAIL PROTECTED];tag=as5c261e75 SIP from address: asterisk sip:[EMAIL PROTECTED] SIP tag: as5c261e75 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via:
Re: [Asterisk-Users] -- MARK --
Well that's what I thought. The syslog had 'asterisk' as the sending process, so I assumed it was a debug statement from Mark S. sigh Mike Isamar Maia wrote: On Mon, 19 Apr 2004, Michael Welter wrote: Every half hour I get -- MARK -- in the syslog. Is this normal behavior? This has nothing to be with asterisk, but with your linux installation. Yes, it is a normal behavior and it is harmless... It is just a half hour stamp to your syslog... I think it was because of MARK Spencer... burn him! :-) Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue out
You can specify a context with single digit extensions for use in a queue, might be in a later cvs release here is the relevant section from the queues.conf ; A context may be specified, in which if the user types a SINGLE ; digit extension while they are in the queue, they will be taken out ; of the queue and sent to that extension in this context. ; ;context = qoutcon Matt On Apr 19, 2004, at 6:03 PM, Jose Maria Guisasola wrote: Please: There is some form so that a user in the queue leaves her (with a digit) and the system execute another command (for example goto a voice mailbox). My version: Asterisk CVS-04/16/04 Thanks in advance -- Jose Mª Guisasola Consultor Técnico CMSI 2002 S.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not working!
I downloaded the cvs development of zaptel... It seems to compile ok, but when ismod I get: lp:/usr/local/src/zaptel# /sbin/insmod wcfxo.o wcfxo.o: unresolved symbol zt_ec_chunk wcfxo.o: unresolved symbol zt_unregister wcfxo.o: unresolved symbol zt_alarm_notify wcfxo.o: unresolved symbol zt_hooksig wcfxo.o: unresolved symbol zt_transmit wcfxo.o: unresolved symbol zt_receive wcfxo.o: unresolved symbol zt_register lp:/usr/local/src/zaptel# lp:/usr/local/src/zaptel# uname -a Linux lp 2.4.24 #3 SMP Thu Jan 15 01:05:12 ART 2004 i686 unknown lp:/usr/local/src/zaptel# lp:/usr/local/src/zaptel# /sbin/insmod -V insmod version 2.4.15 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not working!
On Mon, 19 Apr 2004 [EMAIL PROTECTED] wrote: I downloaded the cvs development of zaptel... It seems to compile ok, but when ismod I get: lp:/usr/local/src/zaptel# /sbin/insmod wcfxo.o wcfxo.o: unresolved symbol zt_ec_chunk wcfxo.o: unresolved symbol zt_unregister wcfxo.o: unresolved symbol zt_alarm_notify wcfxo.o: unresolved symbol zt_hooksig wcfxo.o: unresolved symbol zt_transmit wcfxo.o: unresolved symbol zt_receive wcfxo.o: unresolved symbol zt_register First, 'insmod zaptel'. Then 'insmod wcfxo'. Or, 'modprobe wcfxo'. Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] -- MARK --
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 19 April 2004 10:47 pm, Michael Welter wrote: Well that's what I thought. The syslog had 'asterisk' as the sending process, so I assumed it was a debug statement from Mark S. sigh You guys are funny! Mark Spenser! Haha! I knew immediately it was from Mark Musone! Mike Isamar Maia wrote: On Mon, 19 Apr 2004, Michael Welter wrote: Every half hour I get -- MARK -- in the syslog. Is this normal behavior? This has nothing to be with asterisk, but with your linux installation. Yes, it is a normal behavior and it is harmless... It is just a half hour stamp to your syslog... I think it was because of MARK Spencer... burn him! :-) Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAhJWCljK16xgETzkRAmf+AJ4uFA4axZUBmY6DXCJYBc+R/9MDLgCg45Z7 pxE/MoO1r4gI4QCzXluRFgc= =JySD -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RV: [Asterisk-Users] Not working!
You where SOO right. My idea is to connect the line to my pbx and call from internet (h323?) to my linux box, and then dial an extension.. Is there any doc there? Cause Ive read a few and I did not get much really. I should configure asterisk now and the job is done? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Vic Cross Enviado el: Tuesday, April 20, 2004 12:05 AM Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Not working! On Mon, 19 Apr 2004 [EMAIL PROTECTED] wrote: I downloaded the cvs development of zaptel... It seems to compile ok, but when ismod I get: lp:/usr/local/src/zaptel# /sbin/insmod wcfxo.o wcfxo.o: unresolved symbol zt_ec_chunk wcfxo.o: unresolved symbol zt_unregister wcfxo.o: unresolved symbol zt_alarm_notify wcfxo.o: unresolved symbol zt_hooksig wcfxo.o: unresolved symbol zt_transmit wcfxo.o: unresolved symbol zt_receive wcfxo.o: unresolved symbol zt_register First, 'insmod zaptel'. Then 'insmod wcfxo'. Or, 'modprobe wcfxo'. Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need Help with Dial Plan
Just an update resolved my own issue
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Posted At: Monday, April 19, 2004 9:25 PM
Posted To: Asterisk User Group
Conversation: Need Help with Dial Plan
Subject: [Asterisk-Users] Need Help with Dial Plan
Let me lay it out for you
Call comes in over a T1 - Signal is em_w. The extension is seen as
*callerid*last 4 digits of number being called*. Which is fine in
it self.
I have my extension.conf file set up as follows...
[did]
; Receive call as *calling*called
exten = _.,1,Answer
exten = _.,2,Cut(CALLING=EXTEN,*,2)
exten = _.,3,SetCIDNum(${CALLING})
exten = _.,4,Cut(CALLED=EXTEN,*,3)
exten = _.,5,Goto(main,${CALLED},1)
include = main
[main]
exten = 0031,1,Answer
exten = 0031,2,Goto(TNE-SG,s,1)
Include = did
include = TNE-SG
[TNE-SG]
exten = s,1,Answer
;exten = s,2,agi,tne.agi
exten = s,2,Background(tne-main-thanks)
exten = s,3,Background(tne-main-menu)
exten = 1,1,Goto(default-tne,9100,1)
exten = 2,1,Goto(default-tne,4100,1)
exten = 3,1,Goto(default-tne,4200,1)
exten = 4,1,Goto(default-tne,4300,1)
exten = 5,1,Goto(default-tne,4400,1)
exten = 6,1,Goto(tne-main-menu,s,3)
exten = 7,1,Hangup
include = default-tne
include = main
[default-tne]
include = TNE-SG
; Geoff Clark
exten = 4001,1,Macro(stdexten,4001,SIP/gclark)
;exten = 4001,1,Dial(IAX/home:[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten = 4004,1,Macro(stdexten,4004,SIP/home)
; Kyle Elworthy
exten = 4002,1,Macro(stdexten,4002,SIP/kelworth)
exten = 4003,1,Macro(stdexten,4003,SIP/khome)
; Tech Support Agents
exten = *6,1,AgentCallbackLogin(4001,[EMAIL PROTECTED])
exten = *7,1,AgentCallbackLogin(4002,[EMAIL PROTECTED])
exten = 401,1,Dial(Zap/g1/7046223905)
exten = 402,1,Dial(Zap/g1/7049071514)
exten = 411,1,Answer
exten = 411,2,Wait,2
exten = 411,3,Background(auth-thankyou)
exten = 411,4,Queue(tech-supp)
Where the problem comes in is - I can dial in fine in this scenerio -
but when I go to make an outbound call, it calls the did context and
cut's the call up.
My problem appears to be I need it one way but not the other.. I hope
this makes since...
Thanks,
-gcc
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[Fwd: Re: [Asterisk-Users] IAX config documentation]
Boy after really digging into this, I have discovered that there is more information about each of these topics than I previously realized. Strangely, searching the wiki on iax returns exactly nothing. But searching on iax2 does start to dig up some good stuff. Sorry for the hassle. Tough day. -brian Original Message Subject: Re: [Asterisk-Users] IAX config documentation Date: Mon, 19 Apr 2004 21:22:44 -0400 From: Brian Cuthie [EMAIL PROTECTED] To: [EMAIL PROTECTED] References: [EMAIL PROTECTED] [EMAIL PROTECTED] I know that this stuff is. What I'm looking for is an overview of how these features work in the context of IAX. For instance, trunking is a concept I think we all get. But how do you use IAX to establish trunking between two switches? What's the effect of turning the transfer option on? How are dialplans shared between switches that are connected via IAX? What kinds of authentication are supported? How are keys managed? -brian Steven Critchfield wrote: On Mon, 2004-04-19 at 16:08, Brian Cuthie wrote: Is there any documentation on configuring IAX between * machines? I've noticed references to many topics in the config files, including: - dialplans This is extensions.conf work. Some of it can be shared via the switch command. - trunking Trunking is easy, think of it kind of like a channelized t1. It combines many calls into one packet with call data so as to reduce the overhead of each individual call having it's own resources. Specifically it cuts down on the overhead in IP, and allows you to reclaim some of the bandwidth for more calls. - authentication You do want to know who is trying to call you don't you? - transfers Allows you to get out of the middle of a call. My office loves these as our trunk lines are remote, and when we forward a call out to another trunk line, our local asterisk machine transfers the call back to the machine with trunk lines and removes the VoIP part of the loop. But before I go and try to grok 8000 lines of source (in one file, no less) I was hoping that somewhere there exists even something like a man page that describes the configuration options. There is plenty of banter on the list and info scattered about that google will find for you than reading the source. Of course, you are free to bludgen yourself with the code if you so wish. There's the beginnings of a whitepaper on wiki, but it's self-contradictory in some places, largely incomplete, and just kind of ends abruptly. Yet, it mentions that growing contingent of IAX devices. Huh. Jump in and help finish it when you have read some and start to understand the missing parts. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
