Re: [Asterisk-Users] asterisk to mobile phone
Hello, VoiSmart GSM cards work with Asterisk. Though I have an issue with DTMF detection. See the following pages for details. http://open.voismart.it/index.php/VGSM https://mailman.uli.it/pipermail/visdn-hackers/2006-June/thread.html (Search for DTMF) Regards, Tigran Woodoo People .pGa! wrote: what brand of gsm gateway do you think works well with asterisk? voismart.it - quadgsm ___ --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
Mercie:) how to debug System( script.sh ) Re: [solved] sorry... ; ) Re: [Asterisk-Users] can't run cat $filename inside scripts with system()
Salve Tzafrir! On Sun, 02 Jul 2006, Tzafrir Cohen wrote: #!/bin/bash -x Very good tip! I missed to see what is going on ;) Let me add: to see the debug information of bash -x when run a script with System() call from asterisk, use this at the header of your script: #!/bin/bash -x exec 2/var/log/debug.asterisk.script.name I guess there is no trick to see this inside the asterisk CLI? And probably it is better to write it into an external logfile ;) Mercie, and sunny greetings from Aachen(Europe), rob ___ --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] Re: Digium Hardware Reliability
Also as Bruno suggests I'll pick a new UPS that has the phone line protection as well, though are phone lines are underground to the local station even though we are in a rural location. Cheaper than hanging it on poles I guess. A little tidbit of trivia here I've found the underground lines in some rural areas were a somewhat expensive experiment tried by some telcos. In some rural places in SC it was tried because the strong thunderstorms in the area tended to frequent damage above ground lines. The thought was putting them underground, while a bit more costly, might save some money in the long run. So in certain sections they tried running underground. As a result, those areas of the state usually now can't get things like DSL because it costs them too much to repull the grade of line to support it. That is until they suffer water damage such as in places like Mississippi after the last hurricanes. But I digress... Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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] Best GPL Gui?
I believe vi is GPLed On 6/30/06, Paul Duffy [EMAIL PROTECTED] wrote: Hi Guys With the profusion of different GUI's and Web interfaces out there could someone possibly save me a load of time and let me know which is the best one and why? Also is there an independent site reviewing asterisk GUI's anywhere. I'm looking at Cisco phones and TDM400 and X101P cards. Only GPL versions please. TIA Paul ___ --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] BLINDTRANSFER
http://www.voip-info.org/wiki/view/BLINDTRANSFER On 6/30/06, Kai Ober [EMAIL PROTECTED] wrote: Hi List, i'm fiddling around with a blindtransfers. (and 3PTY) a calls b a transfers b to c (blindtransfer) (c is not a party but a makro which puts b into a MeetMe conference) the conference should be dynamically created. and named after the callerid of a therefor b has to know who which callerid --transfered-- him. is there a VARIABLE or something else, where i can look up WHO transfered b? thx Kai ___ --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] How to continue after a match in an include
I am looking for a way that after a successfully match in one include the next include is still visited. The first include should just set some variables. I tried to number this extension block either with _. or with s and since it matches, the function (setting some variables) have been done. After that, I want to go to the next include, which has a match for _91NNN. However, since the first match was already successful, the next includes are not visited anymore. How can I overcome this problem? bye Ronald Wiplinger ___ --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] channel shows to be in use
when I try asterisk -rx show channels concise I get an output of: SIP/tf.voipmich.com-8671 ... SIP/1110-78ac The phone 1110 is not anymore on a phone call. How can I remove this zombie channel? bye Ronald Wiplinger ___ --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] How to continue after a match in an include
Create a context that is the one that you want the dialplan to visit
last, and dont include this context anywhere, then in the first
context create a line that has a goto statement, something like this:
[default]
include = 1stcontext
[1stcontext]
exten = _.,1,Set(MYVAR=123${EXTEN})
exten = _.,2,Goto(2ndcontext,${EXTEN}),1)
[2ndcontext]
exten = _91NNN,1,Noop(We set this to: ${MYVAR})
if you dial 91222 you should see in the console:
We set this to: 12391222
Hope this helps.
On 7/2/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I am looking for a way that after a successfully match in one include
the next include is still visited.
The first include should just set some variables.
I tried to number this extension block either with _. or with s
and since it matches, the function (setting some variables) have been done.
After that, I want to go to the next include, which has a match for
_91NNN.
However, since the first match was already successful, the next includes
are not visited anymore.
How can I overcome this problem?
bye
Ronald Wiplinger
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Re: [Asterisk-Users] channel shows to be in use
soft hangup command in the CLI will do the magic On 7/2/06, Ronald Wiplinger [EMAIL PROTECTED] wrote: when I try asterisk -rx show channels concise I get an output of: SIP/tf.voipmich.com-8671 ... SIP/1110-78ac The phone 1110 is not anymore on a phone call. How can I remove this zombie channel? bye Ronald Wiplinger ___ --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] dtmfmode=inband but SDP also indicates rfc2833
I'm trying to figure out a way around a problem that I'm having. My carrier sends me a SIP INVITE that indicates that the dtmf modes available are inband (0), and rfc2833 (101). My asterisk server (1.2.9.1) sends back a 200 OK message and shows in its SDP Media Description that we accept inband (0) and rfc2833 (101). My carrier therefore sends all DTMF via rfc2833 which obviously causes problems since asterisk is configured for inband. I've tried going pure rfc2833 with the carrier, and am having DTMF related problems. From the research that I have done with my issue it seems to be a problem with the way asterisk sends the rfc2833 packets out at nearly the same time. Altering the timing that asterisk uses to send the rfc2833 packets seems too deeply seated in asterisk. I therefore have settled on the idea on using inband for dtmf. My termination tests using inband have been successful. So here is what i think will solve my particular problem. I just want to respond with a 200 OK that does not contain anything about rfc2833 in the SDP. Is this fairly doable. I've been diging through chan_sip.c and think I could just make a couple of modifications to make asterisk do what I need it to. I'm hoping to have someone who is familiar with chan_sip.c enlighten me as to whether or not this can be done, and what functions I would need to modify in order to make it happen. Thank You Stagg Shelton www.oneringnetworks.com ___ --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] setting cdr userfield in .call file
Is there a way to set a value for the cdr userfield in a .call file? Like the way you can do: Account: accountnum Anything like Userfield: usernum that I can use? I'm aware that you can use exten = X,s,1,Set(CDR(userfield)=xyz) in the dialplan, but I want the userfield to be filled out even if the call isn't answered and thus doesn't make it into the dialplan. thanks! Cory ___ --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] to.gsm and the.gsm
Can someone send me a link to a GSM sound file (US-English) for the words to and the? BTW - These should be put in the standard asterisk-sounds distribution. I couldn't find them in mine or in the SVN repository. ___ --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] H323 to SIP Gateway
I'm trying to setup an Asterisk box as an H323 to SIP gateway. Basically, I'd like to receive traffic in H323 and forward to another Asterisk box (on the same network) using either IAX2 or SIP so that the second Asterisk box communicates with other gateways using SIP. Therefore, if I receive a request from a remote H323 gateway to dial a particular number, the H323-to-SIP gateway should forward the request to the Asterisk SIP gateway, who would simply terminate the call according to whatever rules are defined in the context. Can anyone tell me how can this be done? I setup chan_oh323 on an * box and played with the configurations but have not been able to make it all work. I can place connect the two * boxes using SIP-to-SIP as well as IAX2-to-IAX2 just fine, but have not gotten the H323 to work. Thanks, Daniel ___ --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] Re: quadBRI in bri_net mode - t3 timer expired
* Paul Hewlett [EMAIL PROTECTED] wrote: On Thursday 29 June 2006 20:08, Sebastian Kayser wrote: i successfully connected our old PBX to an asterisk server with a junghanns quadBRI, the quadBRI ports running in bri_cpe_ptmp mode connected to the interal PBX ISDN ports. Now i tried to turn it round as our PBX depends on it for some features and changed one of the quadBRI ports to bri_net signalling and connected it to one of the external PBX ISDN ports (how do you name that in telco jargon?). did you change the jumpers on the card ? Yes, i did =) Port 3 is jumpered as NT. asterisk*CLI zap show status Description Alarms IRQbpviol CRC4 quadBRI PCI ISDN Card 1 Span 1 [TE] (ca· OK 0 0 0 quadBRI PCI ISDN Card 1 Span 2 [TE] (ca· OK 0 0 0 quadBRI PCI ISDN Card 1 Span 3 [NT] (ca· OK 0 0 0 quadBRI PCI ISDN Card 1 Span 4 [TE] (ca· OK 0 0 0 The card led goes green (indicates an ISDN link). Strange thing here is, the led stays green even if i unplug the cable ... - Sebastian ___ --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] how to ask for number to dial and then dial it?
I want to create an extension say 8000 that prompts the user to
enter a number and then dial that entered number according to a set
of rules. The rules for dialing out are in different context (dial-
out-rules).
[mymenu]
exten = 8000,1,Answer()
[dial-out-rules]
; toll-free numbers out pots line
exten = _1800XXX,1,Dial(${ANALOG_POTS}/${EXTEN})
exten = _1800XXX,n,Hangup()
; long-distance out voip line
exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],30)
exten = _NX,n,Hangup()
etc...
How do I do it?
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Re: [Asterisk-Users] how to ask for number to dial and then dial it?
You could do this:
[mymenu]
exten = 8000,1,Answer()
exten = 8000,n,GoTo(dial-out-rules,s,1)
[dial-out-rules]
; you'll have to record a prompt, or find an appropriate one in the distribution
exten = s,1,Playback(dial_number_after_the_beep)
exten = s,n,Playback(beep)
exten = s,n,WaitExten(5)
; toll-free numbers out pots line
exten = _1800XXX,1,Dial(${ANALOG_POTS}/${EXTEN})
exten = _1800XXX,n,Hangup()
; long-distance out voip line
exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],30)
exten = _NX,n,Hangup()
On 7/2/06, Robert La Ferla [EMAIL PROTECTED] wrote:
I want to create an extension say 8000 that prompts the user to
enter a number and then dial that entered number according to a set
of rules. The rules for dialing out are in different context (dial-
out-rules).
etc...
How do I do it?
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--
web: corybantic.us
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Re: [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833
I answered my own question. My objective was reached with a simple return statement on line 4384 of chan_sip.c in asterisk 1.2.9.1 ftp download. The effect that this has is that asterisk will return a 200 OK that indicates in the SDP that only inband DTMF is supported. My carrier detects this and their NexTone Session Switch sends out dtmf inband. It sucks having to force asterisk to operate in this manner, but hopefully asterisk implementation of rfc2833 will get the bugs worked out, if they are in fact bugs, and not design desicions. Stagg Shelton wrote: I'm trying to figure out a way around a problem that I'm having. My carrier sends me a SIP INVITE that indicates that the dtmf modes available are inband (0), and rfc2833 (101). My asterisk server (1.2.9.1) sends back a 200 OK message and shows in its SDP Media Description that we accept inband (0) and rfc2833 (101). My carrier therefore sends all DTMF via rfc2833 which obviously causes problems since asterisk is configured for inband. I've tried going pure rfc2833 with the carrier, and am having DTMF related problems. From the research that I have done with my issue it seems to be a problem with the way asterisk sends the rfc2833 packets out at nearly the same time. Altering the timing that asterisk uses to send the rfc2833 packets seems too deeply seated in asterisk. I therefore have settled on the idea on using inband for dtmf. My termination tests using inband have been successful. So here is what i think will solve my particular problem. I just want to respond with a 200 OK that does not contain anything about rfc2833 in the SDP. Is this fairly doable. I've been diging through chan_sip.c and think I could just make a couple of modifications to make asterisk do what I need it to. I'm hoping to have someone who is familiar with chan_sip.c enlighten me as to whether or not this can be done, and what functions I would need to modify in order to make it happen. Thank You Stagg Shelton www.oneringnetworks.com ___ --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] dtmfmode=inband but SDP also indicates rfc2833
On Jul 2, 2006, at 1:37 PM, Stagg Shelton wrote: I answered my own question. My objective was reached with a simple return statement on line 4384 of chan_sip.c in asterisk 1.2.9.1 ftp download. The effect that this has is that asterisk will return a 200 OK that indicates in the SDP that only inband DTMF is supported. My carrier detects this and their NexTone Session Switch sends out dtmf inband. It sucks having to force asterisk to operate in this manner, but hopefully asterisk implementation of rfc2833 will get the bugs worked out, if they are in fact bugs, and not design desicions. So a simple: dtmfmode=inband In the appropriate extension (your outbound call terminator) wasn't adequate to force the proper 200 OK message? This doesn't seem like something you should need to change code to do? Glad it's working anyhow, and thanks for sharing your hack. Marty ___ --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: Re: [Asterisk-Users] Sangoma A200 hangup detection
Just wondered if you fixed your hangup problem? Mike On 6/29/06, El Flynn [EMAIL PROTECTED] wrote: chan (Alpha Trilogies Networks) wrote: Hi, Does some one experience the Sangoma A20X-ec series card that cant detect the hangup tone? snip [channels] context = from-pstn3 switchtype = national usecallerid=yes hidecallerid=no transfer=yes echocancel = yes echocancelwhenbridged = yes echotrainning = yes busydetect=yes busycount=1 callprogress=yes relaxdtmf=yes rxgain =-2.5 txgain =-2.5 signalling=fxs_ks group=1 channel=3-4 Any advice? A couple of things: 1. The switchtype setting is only for PRI lines. 2. Try setting callprogress=no, call progress analysis is supposedly only valid in the US. 3. Tune your gain settings until you get an optimal signal level -- google the list or the Wiki, it's quite thoroughly documented. Flynn ___ --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] s / i extension difficulty
I'm having some difficulty with s extensions. For some reason they don't seem to be matching anything. The documentation says that The s extension is used when there is no known called number in the context used. I must be misinterpreting what this means. I take this to mean that if the extension dialed does not match any other extension defined in the context then the s extension is used. So I have. [incoming-sip] exten = s,1,Answer exten = s,n,Playback(enter-ext-of-person) exten = s,n,Hangup exten = 522,1,Answer exten = 522,n,Playback(enter-ext-of-person) exten = 522,n,Hangup If I dial 522 I get the incoming-sip context I hear the playback as expected. If I dial 523 (an extension that isn't matched) I just get a 404 not found, but I expected to get the playback. How is the s extension matched? (asterisk 1.2.9.1) Have I misinterpreted how the s extension is matched? Thanks Tyler ___ --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] s / i extension difficulty
On 6/28/06, Tyler Retzlaff [EMAIL PROTECTED] wrote: I'm having some difficulty with s extensions. For some reason they don't seem to be matching anything. The documentation says that The s extension is used when there is no known called number in the context used. I must be misinterpreting what this means. I take this to mean that if the extension dialed does not match any other extension defined in the context then the s extension is used. No you got it wrong. It means that if the extension dialed is unknow, meaning no extension was dialed but the calles has to handled as in the case with an FXO port that rings, since there is no way to know what extension its trying to reach the s extension is used. If however the extension dialed is known but doesn't exist in the context then the i extension is used. So I have. [incoming-sip] exten = s,1,Answer exten = s,n,Playback(enter-ext-of-person) exten = s,n,Hangup exten = 522,1,Answer exten = 522,n,Playback(enter-ext-of-person) exten = 522,n,Hangup If I dial 522 I get the incoming-sip context I hear the playback as expected. If I dial 523 (an extension that isn't matched) I just get a 404 not found, but I expected to get the playback. How is the s extension matched? (asterisk 1.2.9.1) Have I misinterpreted how the s extension is matched? Thanks Tyler ___ --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] Latest SVN of asterisk-addons doesn't compile
build_tools/mkdep -fPIC -fPIC app_addon_sql_mysql.c app_saycountpl.c cdr_addon_mysql.c res_config_mysql.c app_addon_sql_mysql.c:15:22: error: asterisk.h: No such file or directory app_saycountpl.c:10:22: error: asterisk.h: No such file or directory cdr_addon_mysql.c:22:22: error: asterisk.h: No such file or directory res_config_mysql.c:41:22: error: asterisk.h: No such file or directory gcc -fPIC -fPIC -c -o app_saycountpl.o app_saycountpl.c app_saycountpl.c:10:22: error: asterisk.h: No such file or directory In file included from /usr/include/asterisk/linkedlists.h:23, from /usr/include/asterisk/chanvars.h:26, from /usr/include/asterisk/channel.h:111, from /usr/include/asterisk/file.h:30, from app_saycountpl.c:13: /usr/include/asterisk/lock.h: In function ‘ast_mutex_init’: /usr/include/asterisk/lock.h:534: error: ‘PTHREAD_MUTEX_RECURSIVE’ undeclared (first use in this function) /usr/include/asterisk/lock.h:534: error: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:534: error: for each function it appears in.) $ uname -a Linux localhost 2.6.16-1.2133_FC5 #1 Tue Jun 6 00:52:14 EDT 2006 i686 i686 i386 GNU/Linux $ svn update Fetching external item into 'menuselect' External at revision 17. Fetching external item into 'mxml' External at revision 3. At revision 254. ___ --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] Motorola and Asterisk
Hi GroupDoes anybody knows if Asterisk have plans to work with Motorola and DOCSIS? I'm trying to make work SIP into an PacketCable arquitechture but I can't figure out with Asterisk.ThanksCarlos Bernat ___ --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] Motorola and Asterisk
Isnt DOCSIS a network layer 1 or 2? I TCP/ip would run on top of a DOCSIS network SIP on top of TCP/ip DOCSIS specifies downstream traffic transfer rates between 27 and 36 Mbps over a radio frequency (RF) path in the 50 MHz to 750+ MHz range, and upstream traffic tranfer rates between 320 Kbps and 10 Mbps over a RF path between 5 and 42 MHz. But, because data over cable travels on a shared loop, individuals will see tranfer rates drop as more users gain access So take your DOCSIS standard device and plug it into your * box. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Alberto Bernat Orozco Sent: Sunday, July 02, 2006 9:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Motorola and Asterisk Hi Group Does anybody knows if Asterisk have plans to work with Motorola and DOCSIS? I'm trying to make work SIP into an PacketCable arquitechture but I can't figure out with Asterisk. Thanks Carlos Bernat ___ --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] Re: Digium Hardware Reliability
M.Hockings wrote: Even now, given that I don't know what caused the problem or what solved the problem (for the time being). I might expect that powering the system off may cause software errors due to partially written files but I would NOT expect it to damage the hardware, particularly just a comm card. Hence *my* feeling for *this* card is that it is unreliable. It is however reassuring to hear that overall the reliability of the Digium hardware is good. When you say the card just worked after it apparently went dead, did you switch it around to a different PCI slot? Or did you leave it in the same place, and after some time it worked again? Flynn ___ --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] how to ask for number to dial and then dial it?
Robert La Ferla wrote:
I want to create an extension say 8000 that prompts the user to enter
a number and then dial that entered number according to a set of
rules. The rules for dialing out are in different context (dial-
out-rules).
[some-context]
exten = 8000,1,Playback(please-enter-the-number)
; toll-free numbers out pots line
exten = _1800XXX,1,Dial(${ANALOG_POTS}/${EXTEN})
exten = _1800XXX,n,Hangup()
; long-distance out voip line
exten = _NX,1,Dial(SIP/[EMAIL PROTECTED],30)
exten = _NX,n,Hangup()
exten = i,1,Playback(that-number-is-invalid-ha-ha)
exten = i,2,Congestion
And point the related extensions to [some-context].
Flynn
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Re: [Asterisk-Users] WebPhone
On 06/29/06 04:41 Forrest Beck said the following: Here is a firefox plugin that connects to asterisk via IAX protocol. http://moziax.mozdev.org/ works only on windows, right ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] WebPhone
On 06/29/06 05:17 Tzafrir Cohen said the following: But it's not a web phone by any means. Writing a soft phone in HTML and javascript is practically impossible. with the amount of interest in AJAX, DHTML and the much hyped Web 2.0, this may soon be a possibility as the browsers open up more of their container API. Google Desktop, anyone ? :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Work required - modify Asterisk + SEMS
On 06/29/06 01:18 Jeremy McNamara said the following: why not setup a listen only meetme for the 'listeners' and talk only for the 'talker'? isnt the Page() application used for stuff like this ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] What does it mean?
In CLI and log, I always find the message below. SIP Seeding peer from astdb: '8719' at [EMAIL PROTECTED]:5060 for 60 Can anyone tell me what it means? ___ --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] SIP debug logging
Hi, i am trying to sort out an issue with my SIP provider (I can make outgoing calls but am not recieving calls) and have been trying to use sip debug from the CLI. I am after a way to get these debug messages into a file (I find it easier to go over a file than having to deal with all the re-register messages flying by the screen). I have tried setting the debug field in logger.conf, but this doesn't seem to be doing what I want. Is there an easy way to have these messages going into a file? -- Nikolai Lusan # # # Weblog: http://lusan.id.au/~nikolai/blog # Website:http://lusan.id.au/~nikolai # # ___ --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] WebPhone
On Sat, Jul 01, 2006 at 07:27:37PM +0800, Dinesh Nair wrote: On 06/29/06 04:41 Forrest Beck said the following: Here is a firefox plugin that connects to asterisk via IAX protocol. http://moziax.mozdev.org/ works only on windows, right ? Should also work on Linux. I haven't tested it yet as it requires Firefox 1.5 . You may need iaxclient on your system. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] WebPhone
On Sat, Jul 01, 2006 at 07:28:51PM +0800, Dinesh Nair wrote: On 06/29/06 05:17 Tzafrir Cohen said the following: But it's not a web phone by any means. Writing a soft phone in HTML and javascript is practically impossible. with the amount of interest in AJAX, DHTML and the much hyped Web 2.0, this may soon be a possibility as the browsers open up more of their container API. Google Desktop, anyone ? :) Web pages, evenwith javascript, are still very limited. For instance, they cannot establish UDP communication on their own with other places. An arbitrary TCP connection is also not so trivial. Not to mention some other inherent interface problem. e.g: how do you wait on an event from the server? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Re: Digium Hardware Reliability
El Flynn wrote: M.Hockings wrote: Even now, given that I don't know what caused the problem or what solved the problem (for the time being). I might expect that powering the system off may cause software errors due to partially written files but I would NOT expect it to damage the hardware, particularly just a comm card. Hence *my* feeling for *this* card is that it is unreliable. It is however reassuring to hear that overall the reliability of the Digium hardware is good. When you say the card just worked after it apparently went dead, did you switch it around to a different PCI slot? Or did you leave it in the same place, and after some time it worked again? Flynn I had it in a different slot, out of the machine entirely, in a different slot again then finally back in the original slot. And that is where it is working now. This was a new Digium card in a new IBM (Lenovo) machine. Mike ___ --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] Re: Digium Hardware Reliability
Raymond McKay wrote: Also as Bruno suggests I'll pick a new UPS that has the phone line protection as well, though are phone lines are underground to the local station even though we are in a rural location. Cheaper than hanging it on poles I guess. A little tidbit of trivia here I've found the underground lines in some rural areas were a somewhat expensive experiment tried by some telcos. In some rural places in SC it was tried because the strong thunderstorms in the area tended to frequent damage above ground lines. The thought was putting them underground, while a bit more costly, might save some money in the long run. So in certain sections they tried running underground. As a result, those areas of the state usually now can't get things like DSL because it costs them too much to repull the grade of line to support it. That is until they suffer water damage such as in places like Mississippi after the last hurricanes. But I digress... Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ That is very interesting. I used to have two PSTN lines here but now we use one PSTN and one VOIP nicely unified through Asterisk so people don't really know which they are using. I am unable to get DSL here, so maybe thats a clue as to why. The line we got rid of would not work well in the spring unless you first phoned it from the first line to dry it out or something a bit then you could dial over it. The bad line was an expensive commercial line too that apparently went to a more local switch. The more reliable one that we kept goes to a station about 12 or so km away and is quite noisy which caused me a lot of grief to get the right echo can settings. Mike ___ --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
