Re: [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf
On Thu, 2006-03-02 at 10:59, [EMAIL PROTECTED] wrote: On Thu, 2 Mar 2006, Vladyslav wrote: Just my couple notes on spa3000 and PSTN DTMFs. Such schema: PSTN - SPA3000 - Asterisk Have problems with DTMF detection on incoming calls when call comes from cell phone. Once per 4 times it misdetect some ditigs (whether first digit will be doubled or unrecognized at all). Were tried different (5) cell phones (cell phones providers) Also support from sipura has no clue. I gave up on letting the spa3000 do dtmf and just changed everything to inband, so asterisk handles it now. Are U sure that, when it's set to inband, Asterisk handles DTMFs ? I did set debug and syslog server on sipura to one of my box and grab all debug information from sipura3000. Even when DTMF were set to inband I have seen sipura detected DTMFs from PSTN line. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-3000 and PSTN dtmf
Just my couple notes on spa3000 and PSTN DTMFs. Such schema: PSTN - SPA3000 - Asterisk Have problems with DTMF detection on incoming calls when call comes from cell phone. Once per 4 times it misdetect some ditigs (whether first digit will be doubled or unrecognized at all). Were tried different (5) cell phones (cell phones providers) Also support from sipura has no clue. On Thu, 2006-03-02 at 09:36, Martin Joseph wrote: On Mar 1, 2006, at 3:03 PM, [EMAIL PROTECTED] wrote: On Wed, 1 Mar 2006, Arsen Chaloyan wrote: The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. exactly! There are some other problems with DTMF and spa3000. Sometimes the spa3000 mis-detects remote speaker's voice as DTMF and sends rfc2833 events to asterisk. It's a well known problem: http://voxilla.com/PNphpBB2-viewtopic-t-1054.html It's doesn't seem to be spa3000 specific, as I have seen this same behavior in two other FXO's... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] XLite dtmf issue?
Hi. On Asterisk for Xlite extension U need to set dtmf=inband execute: sip reload and that should be working On Wed, 2006-02-01 at 17:02, Aisling wrote: Hi, Iâm wondering if anyone has experienced an issue with the XLite softphone and asterisk accepting dtmf? I can listen to my voicemail perfectly from my hardphone. However when I dial the voicemail number from my XLite softphone and enter the password at the voicemail prompt, an error appears vm-incorrect and I get an âUnable to read passwordâ message on the asterisk console. Has anyone experienced issues with XLite dtmf? Many thanks, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. __ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Follow ME
Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part accept the call on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration issue. Thank U. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] txfax 18Kb file problem
Hi ALL. I have a problem with TxFax application. (RxFax is working properly) Txfax does not work when sending tiff files bigger than approx. 18Kb. If tiff file smaller than 18Kb everything is OK. Tested in LAN with Panasonic UF-E1 fax machine (tiff files was created by the rxfax from that machine) and SAP-3000. Also tested with some remote fax machines (connected to PSTN and via VoIP). Results are same. Small file in most cases is OK, bigger is FAILED. Environment: Fedora Core 2 and 3, spandsp 0.0.2pre18, libtiff 3.5.7-20.2 and 3.7.1-6, Asterisk 20050427CVS ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Follow Me solution
Better take a look at Dial cmd. and on it's possibility to run Macros. On Thu, 2005-05-19 at 00:19, Ben Johnson wrote: I read an article in the wiki on a (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) that allows Asterisk to forward a call to a cell phone if someone does not answer there office phone. The example waits for the cell phone user to press the # button before bridging the two calls. In the example there is the c switch that tells asterisk to wait for the #. Is there a similar dial statement that would allow me to do this with a IAX2 connection?? Thanks Ben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Connecting 20+ asterisk servers together
Have U tried to use DUNDI for that purpose ?
It's the best solution U could find.
http://www.voip-info.org/wiki-Asterisk+DUNDi+Call+Routing
On Mon, 2005-05-09 at 20:48, Vikram Rangnekar wrote:
+++ Kanuri, Seshu (Company IT) [09/05/05 11:25 -0400]:
Vikram,
Instead of trying to be over-ambitious and try to connect 20 Asterisk
boxes together, why don't you try top connect three (3) of them together
first.
There may lie a plausible solution for you. If this is done, you may go
and string four of them together and so on and so forth.
Take the first step now.
Seshu Kanuri
Hi Seshu,
Three boxes are no problem I already have that running infact 6 boxes to be
exact. But now I have to do 20 plus boxes and they have to be scalable by
that I mean if I want to say 10 more boxes I should be able to do that
easily.
For three boxes I just have IAX trunks between them and have a dialplan
extension for each
exten = _1.,1,Dial(IAX2/${EXTEN:[EMAIL PROTECTED])
exten = _2.,1,Dial(IAX2/${EXTEN:[EMAIL PROTECTED])
exten = _3.,1,Dial(IAX2/${EXTEN:[EMAIL PROTECTED])
I can always use the 'switch' statement to share dialplans too just that the
phones will try to directly send their RTP streams to each other which will
not take advantage of IAX trunking. And if any server is slow then the whole
extension mactching will slow down horribly.
The solution I'm looking for is that say there are 20 servers each can have
their own extensions and whenever a call is placed to an extension thats not
in the local dialplan the asterisk box should sent that call through an IAX
trunk to the asterisk server which contains that extension.
What I understand is that we would need a central database (like enum or
soemthing) to hold all the extensions and the server ip on which that
extension exists. Now when a call is placed that list is lookedup and an IAX
trunk is opened to that server and the call is sent to it. Also if another
call is placed to the same server it uses the same IAX trunk to pass the
second call through too.
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Re: [Asterisk-Users] Fax detected, but no fax extension
In your incoming context add
exten = fax,1,Goto(fax,2202,1)
On Wed, 2005-04-20 at 12:26, Ronald Wiplinger wrote:
-- Starting simple switch on 'Zap/3-1'
-- Executing NoOp(Zap/3-1, 9229443944-) in new stack
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Zapateller(Zap/3-1, ) in new stack
-- Executing BackGround(Zap/3-1, custom/Welcome) in new stack
-- Playing 'custom/Welcome' (language 'default')
Apr 20 16:49:29 NOTICE[29109]: chan_zap.c:4300 zt_read: Fax detected,
but no fax extension
-- Executing BackGround(Zap/3-1, if-u-know-ext-dial) in new stack
-- Playing 'if-u-know-ext-dial' (language 'default')
-- Executing Dial(Zap/3-1,
SIP/601SIP/621ZAP/1r1SIP/610|30|tr) in new stack
-- Called 601
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Called 1r1
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Zap/1-1 is ringing
-- SIP/601-49e2 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- SIP/601-49e2 answered Zap/3-1
-- Hungup 'Zap/1-1'
I tried to use the exension 2201, but it did not work, so I have changed
it to s, but it does not work either.
[fax]
exten = s,1,Macro(faxreceive)
;exten = 2202,1,Macro(faxreceive)
;exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} \
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(3)
How can I solve it?
bye
Ronald
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Re: [Asterisk-Users] help needed for sound device setup
U don't need to have sound device for * sound service running just make sure that you have in modules.conf noload = chan_alsa.so noload = chan_oss.so On Wed, 2005-04-20 at 08:44, [EMAIL PROTECTED] wrote: Hi, I installed asterisk-1-0-7 and running it succesfully. But iam unable to use the sound services. I have the following warning messages when i launch asterisk Apr 19 21:15:40 WARNING[10918]: chan_oss.c:992 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Apr 19 21:15:40 WARNING[10918]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable what i need to do, to install the sound device properly. Does it require any hardware support. i have the following kernel modules for audio support [EMAIL PROTECTED] asterisk-1.0.7]# lsmod | grep audio i810_audio 27720 1 (autoclean) ac97_codec 13640 0 (autoclean) [i810_audio] soundcore 6404 2 (autoclean) [i810_audio] thanks, Somesh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax detected, but no fax extension
You need to add that to context where you have BackGround application
running.
house-day and house-night I believe.
On Wed, 2005-04-20 at 13:16, Ronald Wiplinger wrote:
Vladyslav wrote:
In your incoming context add
exten = fax,1,Goto(fax,2202,1)
It did not work ;-(
[incoming_88097680]
exten = s,1,NoOp(${CALLERIDNUM})
exten = s,2,Wait(1)
exten = s,3,SetCallerId(9${CALLERIDNUM})
exten = s,4,GotoIfTime(08:00-21:20|sun-sat|*|*?house-day,s,1)
exten = s,5,Goto(house-night,s,1)
exten = fax,1,Goto(fax,2201,1)
[fax]
exten = 2201,1,Macro(faxreceive)
;exten = 2202,1,Macro(faxreceive)
;exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} \
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,DBGet(EMAILADDR=extensionemail/${MACRO_EXTEN})
exten = s,3,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(3)
On Wed, 2005-04-20 at 12:26, Ronald Wiplinger wrote:
-- Starting simple switch on 'Zap/3-1'
-- Executing NoOp(Zap/3-1, 9229443944-) in new stack
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Zapateller(Zap/3-1, ) in new stack
-- Executing BackGround(Zap/3-1, custom/Welcome) in new stack
-- Playing 'custom/Welcome' (language 'default')
Apr 20 16:49:29 NOTICE[29109]: chan_zap.c:4300 zt_read: Fax detected,
but no fax extension
-- Executing BackGround(Zap/3-1, if-u-know-ext-dial) in new stack
-- Playing 'if-u-know-ext-dial' (language 'default')
-- Executing Dial(Zap/3-1,
SIP/601SIP/621ZAP/1r1SIP/610|30|tr) in new stack
-- Called 601
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Called 1r1
Apr 20 16:49:43 NOTICE[29109]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3)
-- Zap/1-1 is ringing
-- SIP/601-49e2 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- SIP/601-49e2 answered Zap/3-1
-- Hungup 'Zap/1-1'
I tried to use the exension 2201, but it did not work, so I have changed
it to s, but it does not work either.
[fax]
exten = s,1,Macro(faxreceive)
;exten = 2202,1,Macro(faxreceive)
;exten = 2203,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} \
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
bye
Ronald
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Re: [Asterisk-Users] asterisk on MIPS
I do. On Tue, 2005-04-19 at 02:05, Wang Xiangzhou wrote: Hi, Does anyone run asterisk on MIPS architecture successfully? Thanks, Fox ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgraded now Asterisk won't start
U better add line to your /etc/asterisk/modules.conf noload = chan_phone.so before [global] section (of course in case U don't need that module...) and it will start On Tue, 2005-04-19 at 02:37, Paul A Brown wrote: I have just installed 1.0.7 from the debian source using apt-get. Now it won't run. I try to start it, do a PS and its there for a min then disappears. asterisk -r says can't connect to remote server and asterisk -c says Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = [ Booting.res_odbc loaded. .Registered Config Engine odbc res_config_odbc loaded. .No agent configuration found -- agent support disabled .Unable to load config mgcp.conf, MGCP disabled .Unable to open IAX timing interface: No such file or directory No IAX provisioning configuration found, IAX provisioning disabled. ..Unable to load config skinny.conf, Skinny disabled .Unable to load config phone.conf chan_phone.so: load_module failed, returning -1 Loading module chan_phone.so failed! I can't find any errors in /var/log/asterisk (Not sure which file should show errors but none do anyway) I am bound to of done something daft and apologise in advance. If anyone can help I would appreciate it Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *8 nor *8# works for me!
On Sun, 2005-04-17 at 17:14, Walt Reed wrote: On Fri, Apr 15, 2005 at 10:39:44PM +0800, Ronald Wiplinger said: Eric Wieling wrote: I have put into each phone settings (sip.conf and zapata.conf) in my office: callgroup=1 pickupgroup=1 I cannot pickup any calls from another phone!! What do I miss here? Your SIP phone is eating the *8. You need to look at your SIP phone docs, not Asterisk What am I going to look for, e.g., in a manual for snom 190 and a Budgetone ??? See the Wiki: http://www.voip-info.org/wiki-Asterisk+config+features.conf I had the same problem with Cisco ATA's screwing with the *, so I changed mine to a normal number and everything works great. I never did figure out how to make the cisco pass the *8 properly. For Cisco ATA U just need to modify your dialplan on cisco box add to the beginning of the field DialPlan: *8|... so my cisco dialplan is: *8|*St4-|#St4-|911|1#t8.r9t2-|0#t811.rat4-|^1t4#.- and it picks up remote ext. properly with * bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] App_Conference
I believe you need to modify a little bit member.c file in CVS version they use cid, but in stable version callerid. Just replace properly cid with callerid. It should help with that problem. For example: chan-cid.cid_num change to chan-callerid On Mon, 2005-04-18 at 10:04, E rikje wrote: Anyone tried to build app_conference lately? I'm trying to setup a large conference where i speaker can talk to many listeners, for example 1 speaker and about 100 listeners (who can not speak back to the speaker, 1 way audio only) However, if i try to build app_conference against 1.0.6 or 1.0.7 it won't compile with an error message: make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o app_conference.o app_conference.c gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o conference.o conference.c conference.c: In function `create_conf': conference.c:614: warning: implicit declaration of function `__use_ast_pthread_create_instead__' gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -march=pentium3 -msse -mfpmath=sse,387 -DCRYPTO -DAPP_CONFERENCE_DEBUG -DSILDET=1 -c -o member.o member.c member.c: In function `member_exec': member.c:76: error: structure has no member named `cid' member.c:76: error: structure has no member named `cid' member.c:76: error: structure has no member named `cid' member.c:165: warning: unused variable `ignore_speex_count' make: *** [member.o] Error 1 _ Direct antwoord op je vragen: gebruik MSN Messenger http://messenger.msn.nl/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agents priority in queue
Hello Ppl. Please share info how have you set Agent priority in one queue. Or there is no such kind of thing in current version ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_nv_backgrounddetect - how to make module
On Sun, 2005-03-20 at 23:40, Joseph wrote: How to compile additional module to asterisk? I have app_nv_backgrounddetect.c file and followed instructions below, but make did not generate app_nv_backgrounddetect.so or app_nv_backgrounddetect.o (1) Drop the code in your /usr/src/asterisk/apps directory (2) Edit the Makefile in the apps directory. Add the following line: APPS+=app_nv_backgrounddetect.so (3) Go to /usr/src/asterisk and run make, then run make install I've noticed that in .../apps directory every module has three files file_name.c file_name.o file_name.so How do I get the last two if I have the first one? When U have done first two steps U need just get back to /usr/src/asterisk/ and execute: make that U will have those .so files Everything is pretty clear described on wiki http://voip-info.org/tiki-index.php?page=NVBackgroundDetect ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TxFAX problem
Hi Ppl. Once, couple weeks ago when I have updated * from CVS-HEAD something happen and I could not send a fax anymore. After that I have tried previous * CVS versions with different versions of spandsp (0.0.1, 0.0.2pre4, 0.0.2pre10) but without any changes. I have tried that on Fedora Core 2 with libtiff-3.5.7-16.1 and libtiff-devel-3.5.7-16.1. Everything compiles smoothly, but when I try to send a fax it tries to negotiate and than hangup (on fax machine - incomplete), also tried to send to another fax machine (but result was the same). I get back to spandsp-0.0.1 because that one has at least a bit more debug output than 0.0.2pre10. and here what I got: Slow carrier down Slow carrier up NSF: 20 00 00 79 00 00 00 82 0f 09 03 10 10 00 02 95 80 9c f8 80 40 01 49 02 41 52 41 59 4f 5a 20 55 4b 52 41 49 4e 45 23 20 03 NSF without final frame tag The remote is made by 'Panasonic' DIS: 80 20 ee 99 84 80 11 DIS with final frame tag In state 4 Slow carrier down Slow carrier up XCN: fa XCN with final frame tag In state 4 Disconnecting Changed from phase 3 to 7 Does anyone have a clue what it could be ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TxFAX problem
Thx for your reply. On Wed, 2005-03-16 at 17:35, Steve Underwood wrote: Hi Vladyslav, Use 0.0.2pre1, but add the line fax.verbose = TRUE; just after fax_init(fax, calling_party, NULL); That will turn on the detailed logging. Added, recompiled and tested again. Is the listing you posted the entire log? It looks like there should be more. Yes, before there was some additional information One common mistake people make - Did you use the |caller parameter when running txfax? yes I use that one. Regards, Steve Here is new one : (but it's spandsp-0.0.2pre10) *CLI -- Executing NoOp(SIP/103-dfb6, ) in new stack -- Executing AGI(SIP/103-dfb6, set-timestamp.agi) in new stack -- Executing System(SIP/103-dfb6, echo 16032005-18:15:06 - VladK 103 - SIP/103-dfb6 - 901 /var/log/asterisk/calls) in new stack -- Executing DBput(SIP/103-dfb6, RepeatDial/103=901) in new stack -- DBput: family=RepeatDial, key=103, value=901 -- Executing DBget(SIP/103-dfb6, recv=Record/103) in new stack -- DBget: varname=recv, family=Record, key=103 -- DBget: set variable recv to on -- Executing GotoIf(SIP/103-dfb6, 1?7:9) in new stack -- Goto (from-sip,901,7) -- Executing SetVar(SIP/103-dfb6, CALLFILENAME=20050316-181506-103-901) in new stack -- Executing Monitor(SIP/103-dfb6, wav|20050316-181506-103-901) in new stack -- Executing Goto(SIP/103-dfb6, from-sip-post|901|1) in new stack -- Goto (from-sip-post,901,1) -- Executing Answer(SIP/103-dfb6, ) in new stack -- Executing TxFAX(SIP/103-dfb6, /tmp/testfax.tif|caller) in new stack Slow carrier up Mar 16 18:15:10 NOTICE[10168]: rtp.c:540 ast_rtp_read: Unknown RTP codec 100 received Slow carrier down Slow carrier up Slow carrier down Slow carrier up NSF: 20 00 00 79 00 00 00 82 0f 09 03 10 10 00 02 95 80 9c f8 80 40 01 49 02 41 52 41 59 4f 5a 20 55 4b 52 41 49 4e 45 23 20 03 NSF without final frame tag The remote was made by 'Panasonic' DIS: 80 20 ee 99 84 80 11 DIS with final frame tag In state 10 DIS: V.8 capable Prefer 256 octet blocks Can receive fax Supported data signalling rates: V.27ter, V.29 and V.17 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm or 255mm Recording length: A4 (297mm) and B4 (364mm) Receiver's minimum scan line time: 5ms at 3.85 l/mm: T7.7 = T3.85 Error correction mode R8x15.4lines/mm Metric-based resolution preferred Minimum scan line time for higher resolutions: T15.4 = T7.7 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Start sending document Start tx document Changed from phase 2 to 4 DCS: 83 00 c6 80 80 80 00 HDLC underflow in state 3 Changed from phase 4 to 6 Changed from phase 6 to 3 Slow carrier up CFR: 84 CFR with final frame tag In state 4 Trainability test succeeded Start tx page Slow carrier down Changed from phase 3 to 6 *CLI show ch channel channels channeltypes *CLI show channels Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TxFAX problem
On Wed, 2005-03-16 at 18:25, Steve Underwood wrote: Hi Vladyslav, The log looks good so far. The far end has negotiated. The fast modem has been tested. Transmission of the first page has been. What happens next. I don't think the log really stopped at that point. Did you wait long enough for the page transmission to complete? After that point Changed from phase 3 to 6 fax machine says Incomplete + error code. And that's all. BTW, Fax machine connected via SIPURA-2000 (which registered directly on * and use ulaw) But I could receive fax from PSTN via *-SIPURA-Fax machine Regards, Steve Vladyslav wrote: Here is new one : (but it's spandsp-0.0.2pre10) *CLI -- Executing NoOp(SIP/103-dfb6, ) in new stack -- Executing AGI(SIP/103-dfb6, set-timestamp.agi) in new stack -- Executing System(SIP/103-dfb6, echo 16032005-18:15:06 - VladK 103 - SIP/103-dfb6 - 901 /var/log/asterisk/calls) in new stack -- Executing DBput(SIP/103-dfb6, RepeatDial/103=901) in new stack -- DBput: family=RepeatDial, key=103, value=901 -- Executing DBget(SIP/103-dfb6, recv=Record/103) in new stack -- DBget: varname=recv, family=Record, key=103 -- DBget: set variable recv to on -- Executing GotoIf(SIP/103-dfb6, 1?7:9) in new stack -- Goto (from-sip,901,7) -- Executing SetVar(SIP/103-dfb6, CALLFILENAME=20050316-181506-103-901) in new stack -- Executing Monitor(SIP/103-dfb6, wav|20050316-181506-103-901) in new stack -- Executing Goto(SIP/103-dfb6, from-sip-post|901|1) in new stack -- Goto (from-sip-post,901,1) -- Executing Answer(SIP/103-dfb6, ) in new stack -- Executing TxFAX(SIP/103-dfb6, /tmp/testfax.tif|caller) in new stack Slow carrier up Mar 16 18:15:10 NOTICE[10168]: rtp.c:540 ast_rtp_read: Unknown RTP codec 100 received Slow carrier down Slow carrier up Slow carrier down Slow carrier up NSF: 20 00 00 79 00 00 00 82 0f 09 03 10 10 00 02 95 80 9c f8 80 40 01 49 02 41 52 41 59 4f 5a 20 55 4b 52 41 49 4e 45 23 20 03 NSF without final frame tag The remote was made by 'Panasonic' DIS: 80 20 ee 99 84 80 11 DIS with final frame tag In state 10 DIS: V.8 capable Prefer 256 octet blocks Can receive fax Supported data signalling rates: V.27ter, V.29 and V.17 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm or 255mm Recording length: A4 (297mm) and B4 (364mm) Receiver's minimum scan line time: 5ms at 3.85 l/mm: T7.7 = T3.85 Error correction mode R8x15.4lines/mm Metric-based resolution preferred Minimum scan line time for higher resolutions: T15.4 = T7.7 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Start sending document Start tx document Changed from phase 2 to 4 DCS: 83 00 c6 80 80 80 00 HDLC underflow in state 3 Changed from phase 4 to 6 Changed from phase 6 to 3 Slow carrier up CFR: 84 CFR with final frame tag In state 4 Trainability test succeeded Start tx page Slow carrier down Changed from phase 3 to 6 *CLI show ch channel channels channeltypes *CLI show channels Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Center software opensource or commercial
On Wed, 2005-03-16 at 18:31, Kevin P. Fleming wrote: [EMAIL PROTECTED] wrote: If we want a box that can perform 60 calls. What would be apoproximate budget for that using AMD x86-64 ? 60 calls can easily be done on a 3.4GHz Pentium 4 box, no special hardware is required. Is that with channels recording ? ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA 286 downgrade failure
Good day list, I have a problem with ATA Handy Tone 286. It has been unsuccessfully downgraded via HTTP. Seems like during downgrade there was a problem with connection, because now it's not responding at all. There is no way to get to it's voice menu via phone (by pressing button on it). The button is just flashing ... Could anyone suggest any way to bring that device back to life. Thank U in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Feature automon
There is option automon = *1 in features.conf As I understand when *1 pressed during conversation = recording should begin. But unfortunately it doesn't work for me. I use CVS-HEAD-01/27/05 Does anyone has that feature working? Thanks -- Best Regards VladK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice patch and latest CVS version
Patch could not be applied to the latest cvs version and also http://www.voip-info.org/wiki-Asterisk+Broadvoice+patch?page=Asterisk%20Broadvoice%20patchcomments_threshold=0comments_offset=0comments_sort_mode=commentDate_desccomments_maxComments=10comments_parentId=1209#threadId1210 -- Best regards Vlad ___ 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] Where did USE_MYSQL_FRINDS go ? What to use ?
The documentation is scarce, so could someone please share configs for that. On Fri, 2004-11-26 at 17:47, Brian West wrote: I wonder why people are working on the MySQL specific version if ODBC support is in and being developed. Because people have this misconception that ODBC is slow. bkw ___ 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 -- Best regards Vlad ___ 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] How to transfer value to extensions.conf?
In the call file U could Setvar (4 a channel) and after that use it in
dialplan.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out
On Fri, 2004-11-26 at 18:08, Steven Critchfield wrote:
On Fri, 2004-11-26 at 16:56 +0100, Ning Zhou wrote:
For example, I have a file under /var/spool/asterisk/outgoing, which
include:
channel: zap/g1/12345
MaxRetries: 0
RetryTime: 60
WaitTime: 20
Context: default
Extension:
Priority: 1
And in my extensions.conf file, I have
[default]
exten = ,1,Dial(Zap/g1/34567,20)
exten = ,2,Hangup
Then Asterisk will first dial the 'number 12345, then dial the number
indicted in the exten = which is 34567, then setup a bridge
between them. That is exactly what I want.
My question is, if I like to transfer some variable to the
extensions.conf, to replace the number 34567, since the number dialed
every time is not a fixed phone number, what should I do?
extensions.conf file
[auto-out]
exten = _X.,1,Dial(Zap/g1/${EXTEN}
exten = _X.,2,hangup
Call file
channel: zap/g1/12345
MaxRetries: 0
RetryTime: 60
WaitTime: 20
Context auto-out
Extension: 123456789
Priority: 1
Spend some time learning patern matching.
--
Best regards
Vlad
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RE: [Asterisk-Users] Where did USE_MYSQL_FRINDS go ? What to use ?
so, Guys please anyone provide with an example (config files) how to config sip peers and voicemail in the mysql DB. On Fri, 2004-11-26 at 18:55, Brian West wrote: Because people have this misconception that ODBC is slow. It is not a misconception. It just depends on the application. For a home/small/xyz biz ODBC works fine I guess. The loss in speed is pretty negligible. But I would not want to use ODBC in a large carrier environment. It's slower, another layer with potential bugs etc. Doesn't dance with the KISS principle. At a very large US carrier I was involved in some ODBC tests for their billing system and it went up in flames. If there are ITSP's, VoIP carriers or even large incumbents using ODBC I'd love to hear which application and which ODBC they are using. I don't see how in the world it could go up in flames unless it was pure crap. I had up to 5000 queries(selects) per second on a 1ghz box when I did my testing. No MySQL code will EVER go into asterisk CVS that's the bottom line due to license issues. That's part of the reason I wrote cdr_odbc.c to get around that stupid license issue. bkw ___ 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 -- Best regards Vlad ___ 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] Monitor/Record MeetMe Conversations
Try to mix them and you will get 1 file ... On Thu, 2004-11-11 at 16:40, Matthew Boehm wrote: What is the easiest way to record all parties of a meetme conference into 1 sound file? I tried using Monitor just before the MeetMe call and it gave me files for each person. THanks, Matthew ___ 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 -- Best regards Vlad ___ 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] disable second call / call waiting via SIP
HI! I have a problem with Sjphone on ipaq. It freeze when I receive a call on second line (seems like CPU is not enough). It there a way to restrict call accepting when I'm already on the phone via SIP in *? because: http://www.voip-info.org/wiki-PBX+Call+Waiting For most POTS providers in the United States, Call Waiting may be turned off by dialing *70 before dialing the telephone number. Is there the same in * ? Many thanks in advance. -- Best regards Vlad ___ 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] disable second call / call waiting via SIP
Sorry, have already found that on wiki. http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup On Thu, 2004-10-28 at 13:31, Vladyslav wrote: HI! I have a problem with Sjphone on ipaq. It freeze when I receive a call on second line (seems like CPU is not enough). It there a way to restrict call accepting when I'm already on the phone via SIP in *? because: http://www.voip-info.org/wiki-PBX+Call+Waiting For most POTS providers in the United States, Call Waiting may be turned off by dialing *70 before dialing the telephone number. Is there the same in * ? Many thanks in advance. -- Best regards Vlad ___ 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] 2 * RxFax - TxFax
Hi ALL! I have such schema: Asterisk #1 RxFAX -- Asterisk #2 TxFAX. Both * - almost latest CVS version with spandsp-0.0.2.pre4 on the second * I put call file like this: Channel: hidden with extension specified (in order with second *) Application: txfax Data: /tmp/testfax.tif so asterisks are connected but no sending/receiving on the console I could see established channel and nothing more. Maybe I'm doing something wrong Please advice. -- Best regards Vlad ___ 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] RxFax multiple pages
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten = 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 10,2,Setvar([EMAIL PROTECTED])
exten = 10,3,rxfax(${FAXFILE})
exten = 10,4,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${CALLERID})
mailfax is a program that converts from tiff into jpeg and send a fax to
my email.
When multiple pages were sent I received only the last one.
On the asterisk console I could see that second page is using the same
file name as the first one ( and this is a problem I think).
Does anyone have a success with that ?
--
Best regards
Vlad
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Re: [Asterisk-Users] RxFax multiple pages
Hi.
Thank you all for your replies.
Now I do converting into pdf file and it's ok with multiple pages.
tiff2ps -a ${FAXFILE} | ps2pdf - ${FAXFILE//tiff/pdf}
On Wed, 2004-10-13 at 15:39, Steve Underwood wrote:
Vladyslav wrote:
Hi All.
How to receive multiple pages with rxfax ?
Here is what I have:
exten = 10,1,Setvar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 10,2,Setvar([EMAIL PROTECTED])
exten = 10,3,rxfax(${FAXFILE})
exten = 10,4,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${CALLERID})
mailfax is a program that converts from tiff into jpeg and send a fax to
my email.
When multiple pages were sent I received only the last one.
On the asterisk console I could see that second page is using the same
file name as the first one ( and this is a problem I think).
Does anyone have a success with that ?
You can't properly convert a TIFF file to a JPEG file. JPEG files
contain only one image. TIFF files contain entire documents. If you try
to convert a multi-page TIFF file to a JPEG file the result will depend
on the conversion tool. Most tools are too stupid to do anything
sensible with multi-page TIFFs. You might get the first page, or the
last, or even some random junk. Actually most image handling tools
really suck, and they suck worst when handling TIFF.
Regards,
Steve
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Re: [Asterisk-Users] RxFax - tiff problem
Have your even had success sending couple pages at once without loosing a part of the page? Had the same problem with X100P and it's still unsolved. Just wondering how I could synchronize timing with PSTN on the FXO card. On Thu, 2004-10-07 at 19:41, Snezhana Bekova wrote: Hi! I have another question regarding this issue: If I send multiple pages I do not loose all the remaining pages after a frame slip has occured, but only the rest of the current page. It seems to me as if the frame slip looses some relatively small portion of the image, but not the whole image from this point on to the end of the page. I can receive the next page on the same connection, but I do not receive a portion of the previous page say from where the connection is fine again. Why is this? Isn't it possible to loose some part of a page because of frame slips and still receive the rest of the same page? I am placing a multi-page fax (each page is OK in the beginning) under the following URL: http://sbekova.cyberbg.com/fax/fax-028477011_1.tif Snezhana Bekova Цитат на писмо от Steve Underwood [EMAIL PROTECTED]: Hi, If the problem on the PSTN cside, you need to ensure your E1/T1 is synchronsied to the PSTN. They will not synchronise to you. Look in your zaptel.conf, and ensure it is set to treat your E1/T1 as the primary clock source. Regards, Steve Snezhana Bekova wrote: Hi Steve, thanks very much for your answer! It is unclear to me if this is a clock problem or a network problem? (the traffic comes from some Cisco VoIP equipment at our operator's side and passes a few 100Mbit switches on fiber). If it is a clock problem, what is to be done? Do we need to syncronise the clock of the asterisk machine at our side to its hardware clock in some way (how?) or is the problem at the operator's side? Snezhana Bekova Цитат на писмо от Steve Underwood [EMAIL PROTECTED]: Hi Snezhana, Looking at the audio file, there are frame slips. That is, a clock synchronisation problem, or a machine missing interrupts has caused one or more audio samples to be lost. Fast modems (anything faster than 1200bps basically) cannot cope with even a single lost sample. See http://ww.opencall.org/faq/x26.html and http://ww.opencall.org/faq/x29.html Regards, Steve Snezhana Bekova wrote: Hello, We use tiff version 3.5.7-2 and spandsp-0.0.2pre3 with asterisk version 1.0 on debian unstable. I have posted for our problem wtih fax. RxFax works and we receive faxes, but the tiff file is sometimes malformed although the fax seems to be received correctly. Our asterisk server receives calls and faxes from a mobile operator. We are connected to their Cisco router which connects to our asterisk over VoIP that goes inside a Vlan over 100Mbit fiber which passes a few switches. We move asterisk to other server, so it is powerful enough - it is under load of about 0.01 and CPU load 1% - 2%. So the problem is not the server. Maybe the problem is on the VoIP channel, but we cannot prove it. You can see at http://sbekova.cyberbg.com/fax/ a new example of a received tiff file, some audio log files and debug log. Are the switches and/or routers changing the packet order or is there something else? Thanks in advance. -- Snezhana Bekova ___ 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 ___ 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] video via IAX or SIP
Windows messanger On Thu, 2004-09-23 at 23:47, Florin Andrei wrote: On Thu, 2004-09-23 at 01:59, Vladyslav wrote: HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. What are the clients that you're using? -- Best regards Vlad ___ 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] video via IAX or SIP
HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. from sip.conf [1102] type=friend username=1102 host=dynamic callerid=Veo webcam1102 canreinvite=no disallow=all allow=gsm ;allow=ulaw allow=h261 allow=h263 from iax.conf [peer2] ; 192.168.0.7 type=friend port=4569 auth=md5 secret=second2 context=local host=dynamic qualify=yes trunk=yes jitterbuffer=no disallow=all ;allow=ulaw ;allow=alaw allow=h261 allow=h263 allow=gsm -- IAX2/192.168.0.7:4569/2 answered SIP/1102-62b6 Sep 23 11:49:33 DEBUG[1099414448]: chan_sip.c:825 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found Sep 23 11:49:33 DEBUG[1090837424]: chan_iax2.c:5208 socket_read: Ooh, video format changed to 262144 Sep 23 11:49:33 DEBUG[1112759216]: rtp.c:1166 ast_rtp_write: Ooh, format changed from UNKN to H261 Sep 23 11:49:33 DEBUG[1090837424]: chan_iax2.c:5179 socket_read: Ooh, voice format changed to 2 Sep 23 11:49:33 DEBUG[1112759216]: rtp.c:1166 ast_rtp_write: Ooh, format changed from UNKN to GSM Sep 23 11:50:01 DEBUG[1090837424]: chan_iax2.c:5208 socket_read: Ooh, video format changed to 262144 Sep 23 11:50:04 DEBUG[1112759216]: channel.c:2655 ast_channel_bridge: Didn't get a frame from channel: SIP/1102-62b6 Sep 23 11:50:04 DEBUG[1112759216]: channel.c:2725 ast_channel_bridge: Bridge stops bridging channels SIP/1102-62b6 and IAX2/192.168.0.7:4569/2 Sep 23 11:50:04 DEBUG[1112759216]: chan_iax2.c:2337 iax2_hangup: We're hanging up IAX2/192.168.0.7:4569/2 now... -- Hungup 'IAX2/192.168.0.7:4569/2' Sep 23 11:50:04 DEBUG[1112759216]: app_dial.c:1025 dial_exec: Exiting with DIALSTATUS=ANSWER. == Spawn extension (default, 1101, 102) exited non-zero on 'SIP/1102-62b6 The same situation when I use SIP to dial between two servers. I get Didn't get a frame from channel and hangup. Have tried with different codecs. -- Best regards Vlad ___ 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] X-Lite Meetme problem
On Thu, 2004-09-09 at 09:47, Umar Sear wrote: On Wed, 2004-09-08 at 09:43, Vladyslav wrote: HI! Have a weird problem with X-lite Meetme. When X-Lite user are join to conference room NOT first one, than X-Lite user do not hear anything. This problem gone when X-Lite user get into conference room first (when nobody there). sip.conf [104] context=VoIP-only type=friend username=104 secret=test host=dynamic dtmfmode=rfc2833 mailbox=104 canreinvite=no disallow=all allow=ulaw ;allow=alaw ;allow=gsm On * console have such messages: when X-Lite using ULAW: Sep 8 09:47:17 WARNING[1233853360]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/4) on x-lite ALAW Sep 8 09:49:12 WARNING[1235344304]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/8) on x-lite GSM Sep 8 09:51:50 WARNING[1233804208]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 2 (read/write = 64/2) Please advice. Just a long shot, Try disallowing GSM for [104] Umar It's disallowed in dialplan already. It's commented out ;allow=gsm The problem seems more bigger, because not only such problem with X-lite, but also with Grandstream ATA-286 (image 1.0.5.11). I'm using Asterisk CVS-HEAD-08/04/04 -- Best regards Vlad ___ 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] X-Lite Meetme problem
HI! Have a weird problem with X-lite Meetme. When X-Lite user are join to conference room NOT first one, than X-Lite user do not hear anything. This problem gone when X-Lite user get into conference room first (when nobody there). sip.conf [104] context=VoIP-only type=friend username=104 secret=test host=dynamic dtmfmode=rfc2833 mailbox=104 canreinvite=no disallow=all allow=ulaw ;allow=alaw ;allow=gsm On * console have such messages: when X-Lite using ULAW: Sep 8 09:47:17 WARNING[1233853360]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/4) on x-lite ALAW Sep 8 09:49:12 WARNING[1235344304]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/8) on x-lite GSM Sep 8 09:51:50 WARNING[1233804208]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 2 (read/write = 64/2) Please advice. -- Best regards Vlad ___ 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] X-Lite Meetme problem
And the same problem with Grandstream HandyTone-286 as well On Wed, 2004-09-08 at 11:43, Vladyslav wrote: HI! Have a weird problem with X-lite Meetme. When X-Lite user are join to conference room NOT first one, than X-Lite user do not hear anything. This problem gone when X-Lite user get into conference room first (when nobody there). sip.conf [104] context=VoIP-only type=friend username=104 secret=test host=dynamic dtmfmode=rfc2833 mailbox=104 canreinvite=no disallow=all allow=ulaw ;allow=alaw ;allow=gsm On * console have such messages: when X-Lite using ULAW: Sep 8 09:47:17 WARNING[1233853360]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/4) on x-lite ALAW Sep 8 09:49:12 WARNING[1235344304]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/8) on x-lite GSM Sep 8 09:51:50 WARNING[1233804208]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 2 (read/write = 64/2) Please advice. -- Best regards Vlad ___ 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] cvs server problem
Today morning cvs server checkout problem: cvs server: Updating asterisk-addons/format_mp3 cvs server: failed to create lock directory for `/usr/cvsroot/asterisk-addons/format_mp3' (/usr/cvsroot/asterisk-addons/format_mp3/#cvs.lock): Permission denied cvs server: failed to obtain dir lock in repository `/usr/cvsroot/asterisk-addons/format_mp3' cvs [server aborted]: read lock failed - giving up -- Best regards Vlad ___ 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] video
On Fri, 2004-09-03 at 10:56, Altus Snyman wrote: Good day all I'm interested in video on asterisk using SIP and windows clients Now I did my research on http://www.voip-info.org/wiki-Asterisk+video I have a few question: *On the page they say you need the H.261 H.263? codecs,are these compiled in by default or do I need to do something special and if yes what? They are already in. All U need to do is just allow them in sip.conf *What windows clients are available? Windows messenger 4.7 (In this version U could specify your * server ip) *What cameras/hardware are the best? Have used usb LG PC camera (Flatron) quite good quality 640X480. Please advice and comment on this Thanks Altus ___ 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 -- Best regards Vlad ___ 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] spandsp and certain (e.g. Canon) fax machines
How to force * to write txfax console log into file ? On Thu, 2004-08-26 at 13:24, Steve Underwood wrote: Hi, If that is what you are running, you should be getting audio log files. The have names like /tmp/fax-rx-audio-date and /tmp/fax-tx-audio-date. I need a matching pair and the console log for investigation. Regards, Steve reseaux wrote: Dear Steve the first line of t30.c is - #define LOG_FAX_AUDIO /* * SpanDSP - a series of DSP components for telephony * * t30.c - ITU T.30 FAX transfer processing * * Written by Steve Underwood [EMAIL PROTECTED] * * Copyright (C) 2003 Steve Underwood * -- I think is right uncomment but i dont see ant audio log under /tmp, do you think is possible no audio log? Thanks in advance Dimitri On Thursday 26 August 2004 06:15, Steve Underwood wrote: Hi, You are trying to receive from a Canon FAX machine. The problem I hope the change will fix is in sending *to* a Canon FAX machine. The user who found the bug in spandsp was trying to send to a Philips FAX machine. During negotiation the Philips sent a disconnect message, which is the same problem some people have with some Canon machines. It is not clear from your log why you have problems training the V.29 modem. Can you enable logging, by uncommenting the first line in t30.c, and send me the audio log files you will get in your /tmp directory. Regards, Steve reseaux wrote: Dear Steve i have try the SpanDSP (ver.k and latest Asterisk cvs) with the mod you have write below, but nothing my Canon Fax still dont send the fax: - -- Executing RxFAX(Zap/35-1, /home/user/testfax.tif) in new stack Changed from phase 0 to 1 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 TSI: 43 37 30 38 30 35 33 31 33 33 30 20 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: 0331350807 DCS: 83 00 86 a0 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding 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 Coarse carrier frequency 1728.81 (11) Fast carrier down Fast carrier up Coarse carrier frequency 1699.38 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.06 (4917) Fast carrier down Fast carrier up Coarse carrier frequency 1699.33 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier 1700.09 (4915) Fast carrier down Fast carrier up Coarse carrier frequency 1699.33 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.07 (2924) Fast carrier down -- Executing NoOp(Zap/35-1, DIALEDTIME=) in new stack -- Executing NoOp(Zap/35-1, ANSWEREDTIME=) in new stack -- Executing Hangup(Zap/35-1, ) in new stack I dont know how to debug more, you can give more help to trace the problem? Thanks in advance. Dimitri On Wednesday 25 August 2004 11:34, Steve Underwood wrote: Hi, Several people have reported problems sending faxes from spandsp-0.0.1k to Canon FAX machines. A spandsp user had the same problem with another make of FAX machine, and traced the problem to a bug in the file t30.c of spandsp. Line 542 says s-t4.rx_file[0] where it should say s-t4.tx_file[0]. This fixes his problem, and I suspect it will also fix the Canon fax machine problem. Can someone having problems with Canon machines try this change, and tell me the result? Regards, Steve ___ 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]
[Asterisk-Users] fax output from Asterisk into file
Good day ALL. Could anyone tell me is there a way to get fax debug output into the file when running safe_asterisk ? -- V.8 capable Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter, V.29 and V.17 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm or 255mm Recording length: A4 (297mm) and B4 (364mm) Receiver's minimum scan line time: 5ms at 3.85 l/mm: T7.7 = T3.85 Error correction mode R8x15.4lines/mm OK Metric-based resolution preferred Minimum scan line time for higher resolutions: T15.4 = T7.7 DCS: Selected data signalling rate: V.29, 9600bps The thing is that I'd like to make a web page where I'm uploading a file (it's being converted into tiff), creates a call file in /var/spool/asterisk/outgoing and send a fax. I have such call file: Channel: TRUNK/[EMAIL PROTECTED] Callerid: 103 Application: txfax Data: /tmp/fax2send.tiff But there is a problem with status of the fax transmit. When another end pickup a line and even no fax sent it hangups and * delete call file in the outgoing folder. -- Best regards Vlad ___ 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] iconnect inbound - so do we know how to fix it
Try to comment out in your sip.conf ;qualify=yes On Mon, 2004-08-09 at 06:52, Sathya Weerasooriya wrote: Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -- Best regards Vlad ___ 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] rxfax killed asterisk
HI All. I'm using tiff-v.3.5.7 and spandsp-0.0.1k with latest * cvs on Slackware-10.0. Here is debug messages from * console. Please advise. Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1699.96 (66) Training error 2.228910 Training succeeded (constellation mismatch 4.905620) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 2 Start rx page Killed P.S. Have the same installation on Fedora Core 2 and everything works ok. But I need it on Slackware :) -- Best regards Vlad ___ 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] faxing
BTW, compilation of rxfax with latest CVS-2004-07-29 fails. and Makefile.patch (which is on the site) should be modified as well. gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c In file included from app_rxfax.c:14: ../include/asterisk/lock.h: In function `ast_mutex_init': ../include/asterisk/lock.h:300: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) ../include/asterisk/lock.h:300: (Each undeclared identifier is reported only once ../include/asterisk/lock.h:300: for each function it appears in.) make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/install/cvs/2004-07-29/asterisk/apps' make: *** [subdirs] Error 1 On Fri, 2004-07-30 at 05:45, Steve Underwood wrote: Wrong way around. It is passive mode which is giving trouble. I need to fix the firewalling. Active mode should be OK right now. Regards, Steve -- Best regards Vlad ___ 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] faxing
What's wrong with ftp://ftp.opencall.org/pub/ It says Can't open data connection On Thu, 2004-07-29 at 17:44, Steve Underwood wrote: [EMAIL PROTECTED] wrote: What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers. Outgoing has been bad. It seems to work the best if the Sipura user agents have echo cancelation off, but we have to have echo cancelation on our outbound gateways or there is echo in the voice path. Faxing outbound works very rarely, and if it does, it usually can only send a page or two before we get the infamous Poor line condition. Does anyone have a suitable FAX setup working? Using G.711 u or A may work, but don't count on it. Take a look at http://www.opencall.org/faq Regards, Steve ___ 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 -- Best regards Vlad ___ 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] *8# into invalid extensions
ok. Thank U for a hint. I have find out, the problem was with my ATA-186. That box just use '#' not as sending key. Does anyone know how to force ATA-186 to use '#' as sending key. Have tried *8 from softphone and that works fine. On Mon, 2004-07-05 at 21:19, Brancaleoni Matteo wrote: Hi Il lun, 2004-07-05 alle 20:12, Brian K. West ha scritto: *8# works on sip that uses the # as the send key. sure, but since he gets -- Sent into invalid extension '*8#' in context 'from-sip-post'... means that he's sending *8# ... matteo -- Best regards Vlad ___ 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] *8# into invalid extensions
Hi All! Have a problem with remote call pickup via sip. When 1 sip phone is ringing and I'm trying to pickup a call from another sip phone by dialing *8# I'm getting: -- Sent into invalid extension '*8#' in context 'from-sip-post' on SIP/ciscok-8d39 such configs: zapata.conf -- context=inbound-analog callgroup=2 channel=2 -- sip.conf -- [ciscok] type=friend host=dynamic username=ciscok canreinvite=no callgroup=2 pickupgroup=2 mailbox=100 qualify=1000 dtmfmode=rfc2833 trunk=yes [ciscok2] type=friend host=dynamic username=ciscok2 canreinvite=no callgroup=2 pickupgroup=2 qualify=1000 dtmfmode=rfc2833 trunk=yes -- Please help. -- Best regards Vlad ___ 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] Grandstream HT286 1.0.4.63 Meetme
Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec = it connects to conference room, but hear nothing (just dead ear). 2. When HT use G729 codec = it gets busy signal and I could see such output on asterisk console ( Jul 1 07:26:14 WARNING[737298]: chan_sip.c:1611 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/256) Jul 1 07:26:14 WARNING[737298]: app_meetme.c:896 conf_run: Unable to write frame to channel: No child processes Jul 1 07:26:14 WARNING[737298]: app_meetme.c:896 conf_run: Unable to write frame to channel: No child processes ) 3. When HT use G711 (ULAW) = it gets into conference room without any problem. Any advise appreciated. -- Best regards Vlad ___ 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] Grandstream HT286 1.0.4.63 Meetme
On Fri, 2004-07-02 at 09:22, Dave Cotton wrote: 2. When HT use G729 codec = it gets busy signal and I could see such output on asterisk console ( Jul 1 07:26:14 WARNING[737298]: chan_sip.c:1611 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 4/256) Jul 1 07:26:14 WARNING[737298]: app_meetme.c:896 conf_run: Unable to write frame to channel: No child processes Jul 1 07:26:14 WARNING[737298]: app_meetme.c:896 conf_run: Unable to write frame to channel: No child processes ) Asterisk does _not_ support G729 unless you have the license paid for and installed. I have two licenses installed on my asterisk box. 3. When HT use G711 (ULAW) = it gets into conference room without any problem. Any advise appreciated. Use Ulaw :) -- Best regards Vlad ___ 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] pulse dialing
Good day, does anyone have pulse dialing working ? http://voip-info.org/tiki-index.php?page=Asterisk+zaptel+pulse+dialing At the link above there is a statement: configuration for European telephone lines will look like: make_time=63 break_time=37 pause_time=800 So where these pamameters should go to ? zapata.conf ignore them. I have tried on both version of asterisk (cvs and stable) with related patches (pulsepatchcvs.txt and pulsepatch1.0.txt). Patches applied well, however * pulse dialing doesn't work properly. I receive unusual busy signal. -- Best regards Vlad ___ 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] dialplan AGI DTMF
Good day All. Is there a way to pass DTMF signals to AGI script during conversation ? Actually here what I want to make: Users are usually dial using dialplan and when someone press *4 (during conversation) I want to have agi script to deal with that, but those users should keep talking and even didn't notice that one of them press something. Is there a way to do that or it's complete nonsense? -- Best regards Vlad ___ 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] RxFAX generates no tiff file
Good day. I have such problem with rxfax: When I send a fax in fine mode resolution i receive only 50-60% (sometimes 25%) of the page and the rest is compressed lines. http://robik.azhelp.net/1084284449.0.tif http://robik.azhelp.net/1084289786.1.tif Please advise On Mon, 2004-05-24 at 03:03, Steve Underwood wrote: For most people who are sure they have no frame slips, the problem usually turns out to be frame slips :-) If you are *really* sure you do not have frame slips, then uncomment the first line in t30.c, and rebuild and reinstall spandsp. The when you exchange a fax you should end up with a pair of audio files in your /tmp directory - one for the transmit signal and one for the receive signal. Send those to me, and I will investigate. Regards, Steve -- Best regards Vlad ___ 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] fine mode receive fax problem
Hi, ALL. Have a problem with tiff image when receive fax in fine mode via Zap (FXO card). The same via SIP is fine. Could receive faxes in standard resolution without a problem, but fine or super fine mode got tiff images corrupted. With fine resolution, simply have twice the lines and it looks like when sending via Zap it get only before predefined amount of lines and the rest simply skips. Any thoughts ? -- Best regards Vlad ___ 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
